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-rw-r--r--config/freeswitch/dialplan.default.xml673
-rw-r--r--config/freeswitch/dialplan.public.xml69
-rwxr-xr-xconfig/freeswitch/libncurses.so.5.7bin0 -> 127748 bytes
3 files changed, 742 insertions, 0 deletions
diff --git a/config/freeswitch/dialplan.default.xml b/config/freeswitch/dialplan.default.xml
new file mode 100644
index 00000000..04a31950
--- /dev/null
+++ b/config/freeswitch/dialplan.default.xml
@@ -0,0 +1,673 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+ NOTICE:
+
+ This context is usually accessed via authenticated callers on the sip profile on port 5060
+ or transfered callers from the public context which arrived via the sip profile on port 5080.
+
+ Authenticated users will use the user_context variable on the user to determine what context
+ they can access. You can also add a user in the directory with the cidr= attribute acl.conf.xml
+ will build the domains acl using this value.
+-->
+<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
+<include>
+ <context name="default">
+
+ <extension name="unloop">
+ <condition field="${unroll_loops}" expression="^true$"/>
+ <condition field="${sip_looped_call}" expression="^true$">
+ <action application="deflect" data="${destination_number}"/>
+ </condition>
+ </extension>
+
+ <!-- Example of doing things based on time of day. -->
+ <extension name="tod_example" continue="true">
+ <!-- man strftime - M-F, 9AM to 6PM -->
+ <condition field="${strftime(%w)}" expression="^([1-5])$"/>
+ <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$">
+ <action application="set" data="open=true"/>
+ </condition>
+ </extension>
+
+ <extension name="global-intercept">
+ <condition field="destination_number" expression="^\*886$">
+ <action application="answer"/>
+ <action application="intercept" data="${hash(select/${domain_name}-last_dial/global)}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="group-intercept">
+ <condition field="destination_number" expression="^\*8$">
+ <action application="answer"/>
+ <action application="intercept" data="${hash(select/${domain_name}-last_dial/${callgroup})}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="intercept-ext">
+ <condition field="destination_number" expression="^\*\*(\d+)$">
+ <action application="answer"/>
+ <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/>
+ <action application="sleep" data="2000"/>
+ </condition>
+ </extension>
+
+ <extension name="redial">
+ <condition field="destination_number" expression="^\*870$">
+ <action application="transfer" data="${hash(select/${domain_name}-last_dial/${caller_id_number})}"/>
+ </condition>
+ </extension>
+
+ <extension name="global" continue="true">
+ <condition field="${network_addr}" expression="^$" break="never">
+ <action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/>
+ <anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/>
+ </condition>
+ <condition field="${call_debug}" expression="^true$" break="never">
+ <action application="info"/>
+ </condition>
+ <!--
+ This is an example of how to auto detect if telephone-event is missing and activate inband detection
+ -->
+ <!--
+ <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
+ <action application="set" data="rtp_payload_number=$1"/>
+ <anti-action application="start_dtmf"/>
+ </condition>
+ -->
+ <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
+ <action application="set" data="sip_secure_media=true"/>
+ <!-- Offer SRTP on outbound legs if we have it on inbound. -->
+ <!-- <action application="export" data="sip_secure_media=true"/> -->
+ </condition>
+
+ <condition>
+ <action application="hash" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
+ <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
+ <action application="hash" data="insert/${domain_name}-last_dial/global/${uuid}"/>
+ </condition>
+ </extension>
+
+ <!-- If sip_req_host is not a local domain then this has to be an external sip uri -->
+ <!--
+ <extension name="external_sip_uri" continue="true">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="${outside_call}" expression="^$"/>
+ <condition field="${domain_exists(${sip_req_host})}" expression="true">
+ <anti-action application="bridge" data="sofia/${use_profile}/${sip_to_uri}"/>
+ </condition>
+ </extension>
+ -->
+ <!--
+ snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference
+ -->
+
+ <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>
+
+ <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="eavesdrop">
+ <condition field="destination_number" expression="^\*88(.*)$|^\*0(.*)$">
+ <action application="answer"/>
+ <action application="eavesdrop" data="${hash(select/${domain_name}-spymap/$1)}"/>
+ </condition>
+ </extension>
+
+ <extension name="eavesdrop">
+ <condition field="destination_number" expression="^\*779$">
+ <action application="answer"/>
+ <action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/>
+ <action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/>
+ <action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/>
+ <action application="eavesdrop" data="all"/>
+ </condition>
+ </extension>
+
+ <extension name="call_return">
+ <condition field="destination_number" expression="^\*69$|^869$|^lcr$">
+ <action application="transfer" data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/>
+ </condition>
+ </extension>
+
+ <extension name="del-group">
+ <condition field="destination_number" expression="^\*80(\d{2})$">
+ <action application="answer"/>
+ <action application="group" data="delete:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/>
+ <action application="gentones" data="%(1000, 0, 320)"/>
+ </condition>
+ </extension>
+
+ <extension name="add-group">
+ <condition field="destination_number" expression="^\*81(\d{2})$">
+ <action application="answer"/>
+ <action application="group" data="insert:$1@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}"/>
+ <action application="gentones" data="%(1000, 0, 640)"/>
+ </condition>
+ </extension>
+
+ <extension name="call-group-simo">
+ <condition field="destination_number" expression="^\*82(\d{2})$">
+ <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/>
+ </condition>
+ </extension>
+
+ <extension name="call-group-order">
+ <condition field="destination_number" expression="^\*83(\d{2})$">
+ <action application="set" data="call_timeout=10"/>
+ <action application="bridge" data="{ignore_early_media=true}${group(call:$1@${domain_name}:order)}"/>
+ </condition>
+ </extension>
+
+ <extension name="extension-intercom">
+ <condition field="destination_number" expression="^\*8(10[01][0-9])$">
+ <action application="set" data="dialed_extension=$1"/>
+ <action application="export" data="sip_auto_answer=true"/>
+ <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
+ </condition>
+ </extension>
+
+ <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
+
+ <!--
+ dial the extension (1000-1019) for 30 seconds and go to voicemail if the
+ call fails (continue_on_fail=true), otherwise hang up after a successful
+ bridge (hangup_after-bridge=true)
+ -->
+ <extension name="Local_Extension">
+ <condition field="destination_number" expression="^(\d{4})$|^(\d{3})$">
+ <action application="set" data="dialed_extension=$1"/>
+ <action application="export" data="dialed_extension=$1"/>
+ <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
+ <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
+ <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+ <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
+ <action application="set" data="ringback=${us-ring}"/>
+ <action application="set" data="transfer_ringback=$${hold_music}"/>
+ <action application="set" data="call_timeout=30"/>
+ <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
+ <action application="set" data="hangup_after_bridge=true"/>
+ <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
+ <action application="set" data="continue_on_fail=true"/>
+ <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
+ <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
+ <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
+ <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
+ <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
+ <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
+ </condition>
+ </extension>
+
+ <extension name="group_dial_sales">
+ <condition field="destination_number" expression="^\*2000$">
+ <action application="bridge" data="${group_call(sales@${domain_name})}"/>
+ </condition>
+ </extension>
+
+ <extension name="group_dial_support">
+ <condition field="destination_number" expression="^\*2001$">
+ <action application="bridge" data="group/support@${domain_name}"/>
+ </condition>
+ </extension>
+
+ <extension name="group_dial_billing">
+ <condition field="destination_number" expression="^\*2002$">
+ <action application="bridge" data="group/billing@${domain_name}"/>
+ </condition>
+ </extension>
+
+ <!-- voicemail operator extension -->
+ <extension name="operator">
+ <condition field="destination_number" expression="^\*operator$|^0$">
+ <action application="set" data="transfer_ringback=$${hold_music}"/>
+ <action application="transfer" data="1000 XML features"/>
+ </condition>
+ </extension>
+
+ <!-- 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>
+
+ <!-- dial via SIP uri -->
+ <extension name="sip_uri">
+ <condition field="destination_number" expression="^sip:(.*)$">
+ <action application="bridge" data="sofia/${use_profile}/$1"/>
+ </condition>
+ </extension>
+
+ <!--
+ 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>
+
+ <!-- dial the freeswitch conference via SIP-->
+ <extension name="freeswitch_public_conf_via_sip">
+ <condition field="destination_number" expression="^\*9(888|1616|3232)$">
+ <action application="export" data="hold_music=silence"/>
+ <!--
+ This will take the SAS from the b-leg and send it to the display on the a-leg phone.
+ Known working with Polycom and Snom maybe others.
+ -->
+ <!--
+ <action application="set" data="exec_after_bridge_app=${sched_api(+4 zrtp expand uuid_display ${uuid} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas1_string )} \${uuid_getvar(\${uuid_getvar(${uuid} signal_bond)} zrtp_sas2_string )} )}"/>
+ <action application="export" data="nolocal:zrtp_secure_media=true"/>
+ -->
+ <action application="bridge" data="sofia/${use_profile}/$1@conference.freeswitch.org"/>
+ </condition>
+ </extension>
+
+ <!--
+ This extension will start a conference and invite a group.
+ At anytime the participant can dial *2 to bridge directly to the boss.
+ All other callers are then hung up on.
+ -->
+ <extension name="mad_boss_intercom">
+ <condition field="destination_number" expression="^\*0911$">
+ <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/>
+ <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
+ <action application="set" data="conference_auto_outcall_timeout=60"/>
+ <action application="set" data="conference_auto_outcall_flags=mute"/>
+ <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
+ <action application="set" data="sip_exclude_contact=${network_addr}"/>
+ <action application="conference_set_auto_outcall" data="${group_call(sales)}"/>
+ <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/>
+ </condition>
+ </extension>
+
+ <!--
+ This extension will start a conference and invite a few of people.
+ At anytime the participant can dial *2 to bridge directly to the boss.
+ All other callers are then hung up on.
+ -->
+ <extension name="mad_boss_intercom">
+ <condition field="destination_number" expression="^\*0912$">
+ <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss2"/>
+ <action application="set" data="conference_auto_outcall_caller_id_number=0912"/>
+ <action application="set" data="conference_auto_outcall_timeout=60"/>
+ <action application="set" data="conference_auto_outcall_flags=mute"/>
+ <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
+ <action application="set" data="sip_exclude_contact=${network_addr}"/>
+ <action application="conference_set_auto_outcall" data="loopback/9999"/>
+ <action application="conference" data="madboss_intercom2@default+flags{endconf|deaf}"/>
+ </condition>
+ </extension>
+
+ <!--This extension will start a conference and invite several people upon entering -->
+ <extension name="mad_boss">
+ <condition field="destination_number" expression="^\*0913$">
+ <!--These params effect the outcalls made once you join-->
+ <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss"/>
+ <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
+ <action application="set" data="conference_auto_outcall_timeout=60"/>
+ <action application="set" data="conference_auto_outcall_flags=none"/>
+ <!--<action application="set" data="conference_auto_outcall_announce=say:You have been called into an emergency conference"/>-->
+ <!--Add as many of these as you need, These are the people you are going to call-->
+ <action application="conference_set_auto_outcall" data="loopback/9999"/>
+ <action application="conference" data="madboss3@default"/>
+ </condition>
+ </extension>
+
+ <!-- a sample IVR -->
+ <extension name="ivr_demo">
+ <condition field="destination_number" expression="^\*5000$">
+ <action application="answer"/>
+ <action application="sleep" data="2000"/>
+ <action application="ivr" data="demo_ivr"/>
+ </condition>
+ </extension>
+
+ <!-- Create a conference on the fly and pull someone in at the same time. -->
+ <extension name="dynamic_conference">
+ <condition field="destination_number" expression="^\*5001$">
+ <action application="conference" data="bridge:mydynaconf:sofia/${use_profile}/1234@conference.freeswitch.org"/>
+ </condition>
+ </extension>
+
+ <extension name="rtp_multicast_page">
+ <condition field="destination_number" expression="^\*pagegroup$|^\*7243">
+ <action application="answer"/>
+ <action application="esf_page_group"/>
+ </condition>
+ </extension>
+
+ <!--
+ 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>
+
+ <!--
+ This extension is used with snom phones.
+
+ Set a function key to park+lot (lot being a number or name.)
+ Set type to Park+Orbit. You can then park and pickup using
+ the softkey on the phone. Should work with other phones.
+ -->
+ <extension name="park">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="park\+(\d+)">
+ <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
+ </condition>
+ </extension>
+ <!--
+ The extension is parking pickup with a to param of the fifo we are calling
+ Some phones send things like orbit= and you can extract that info.
+ -->
+ <extension name="unpark">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="^parking$"/>
+ <condition field="${sip_to_params}" expression="fifo\=(\d+)">
+ <action application="answer"/>
+ <action application="fifo" data="$1@${domain_name} out nowait"/>
+ </condition>
+ </extension>
+
+ <!--
+ This extension is used with linksys phones.
+
+ Set a Phone tab option Call Park Serv to yes. You can park and
+ pickup using soft keys "park" and "unpark" found during
+ active call when moving navigation button. The other option
+ is to use phone's star codes (defaults to *38 and *39).
+ -->
+ <extension name="park">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="callpark"/>
+ <condition field="${sip_refer_to}">
+ <expression><![CDATA[<sip:callpark@${domain_name};orbit=(\d+)>]]></expression>
+ <action application="fifo" data="$1@${domain_name} in undef $${hold_music}"/>
+ </condition>
+ </extension>
+
+ <!--
+ This extension is used with linksys phones.
+
+ The extension is parking pickup with a to param of the fifo
+ we are calling. Linksys sends orbit=<parkingslotnumber>
+ and we extract that info.
+ -->
+ <extension name="unpark">
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="pickup"/>
+ <condition field="${sip_to_params}" expression="orbit\=(\d+)">
+ <action application="answer"/>
+ <action application="fifo" data="$1@${domain_name} out nowait"/>
+ </condition>
+ </extension>
+
+ <!--
+ Here are some examples of how to override the ringback heard by the
+ far end. You have two variables that you can use to override this.
+
+ ringback - used when a call isn't answered. (early media)
+ transfer_ringback - used when the call is already answered. (post answer)
+ -->
+
+ <!-- Demonstration of how to override the ringback in various situations -->
+ <extension name="wait">
+ <condition field="destination_number" expression="^wait$">
+ <action application="pre_answer"/>
+ <action application="sleep" data="20000"/>
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="playback" data="voicemail/vm-goodbye.wav"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <extension name="fax_receive">
+ <condition field="destination_number" expression="^\*9978$">
+ <action application="answer" />
+ <action application="playback" data="silence_stream://2000"/>
+ <action application="rxfax" data="/tmp/rxfax.tif"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <extension name="fax_transmit">
+ <condition field="destination_number" expression="^\*9979$">
+ <action application="txfax" data="/tmp/txfax.tif"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <!-- Send a 180 and let the far end generate ringback. -->
+ <extension name="ringback_180">
+ <condition field="destination_number" expression="^\*9980$">
+ <action application="ring_ready"/>
+ <action application="sleep" data="20000"/>
+ <action application="answer"/>
+ <action application="sleep" data="1000"/>
+ <action application="playback" data="voicemail/vm-goodbye.wav"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <!-- Send a 183 and send uk-ring as the ringtone. (early media) -->
+ <extension name="ringback_183_uk_ring">
+ <condition field="destination_number" expression="^\*9981$">
+ <action application="set" data="ringback=$${uk-ring}"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <!-- Send a 183 and use music as the ringtone. (early media) -->
+ <extension name="ringback_183_music_ring">
+ <condition field="destination_number" expression="^\*9982$">
+ <action application="set" data="ringback=$${hold_music}"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <!-- Answer the call and use music as the ringtone. (post answer) -->
+ <extension name="ringback_post_answer_uk_ring">
+ <condition field="destination_number" expression="^\*9983$">
+ <action application="set" data="transfer_ringback=$${uk-ring}"/>
+ <action application="answer"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <!-- Answer the call and use music as the ringtone. (post answer) -->
+ <extension name="ringback_post_answer_music">
+ <condition field="destination_number" expression="^\*9984$">
+ <action application="set" data="transfer_ringback=$${hold_music}"/>
+ <action application="answer"/>
+ <action application="bridge" data="loopback/wait"/>
+ </condition>
+ </extension>
+
+ <extension name="ClueCon">
+ <condition field="destination_number" expression="^\*9991$">
+ <action application="set" data="effective_caller_id_name=ClueCon"/>
+ <action application="bridge" data="sofia/$${domain}/brian@bkw.org"/>
+ </condition>
+ </extension>
+
+ <extension name="show_info">
+ <condition field="destination_number" expression="^\*9992$">
+ <action application="answer"/>
+ <action application="info"/>
+ <action application="sleep" data="250"/>
+ <action application="hangup"/>
+ </condition>
+ </extension>
+
+ <extension name="video_record">
+ <condition field="destination_number" expression="^\*9993$">
+ <action application="answer"/>
+ <action application="record_fsv" data="/tmp/testrecord.fsv"/>
+ </condition>
+ </extension>
+
+ <extension name="video_playback">
+ <condition field="destination_number" expression="^\*9994$">
+ <action application="answer"/>
+ <action application="play_fsv" data="/tmp/testrecord.fsv"/>
+ </condition>
+ </extension>
+
+ <extension name="delay_echo">
+ <condition field="destination_number" expression="^\*9995$">
+ <action application="answer"/>
+ <action application="delay_echo" data="5000"/>
+ </condition>
+ </extension>
+
+ <extension name="echo">
+ <condition field="destination_number" expression="^\*9996$">
+ <action application="answer"/>
+ <action application="echo"/>
+ </condition>
+ </extension>
+
+ <extension name="milliwatt">
+ <condition field="destination_number" expression="^\*9997$">
+ <action application="answer"/>
+ <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/>
+ </condition>
+ </extension>
+
+ <extension name="tone_stream">
+ <condition field="destination_number" expression="^\*9998$">
+ <action application="answer"/>
+ <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/>
+ </condition>
+ </extension>
+
+ <!--
+ You will no longer hear the bong tone. The wav file is playing stating the call is secure.
+ The file will not play unless you have both TLS and SRTP active.
+ -->
+
+ <extension name="hold_music">
+ <condition field="destination_number" expression="^\*9999$"/>
+ <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
+ <action application="answer"/>
+ <action application="execute_extension" data="is_secure XML features"/>
+ <action application="playback" data="$${hold_music}"/>
+ <!-- This really should be an IVR for zrtp enrollment but this is just a demo-->
+ <anti-action application="set" data="zrtp_enrollment=true"/>
+ <anti-action application="answer"/>
+ <anti-action application="playback" data="$${hold_music}"/>
+ </condition>
+ </extension>
+
+ <!--
+ You can place files in the default directory to get included.
+ -->
+ <!--<X-PRE-PROCESS cmd="include" data="default/*.xml"/>-->
+
+ <!--
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+
+ Anything you put below this line will usually get ignored due to the file in
+ default/99999_enum.xml as it will transfer the call to the enum dialplan.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+ -->
+
+ <extension name="enum">
+ <condition field="${module_exists(mod_enum)}" expression="true"/>
+ <condition field="destination_number" expression="^(.*)$">
+ <action application="transfer" data="$1 enum"/>
+ </condition>
+ </extension>
+
+ <!--
+ <extension name="refer">
+ <condition field="${sip_refer_to}">
+ <expression><![CDATA[<sip:${destination_number}@${domain_name}>]]></expression>
+ </condition>
+ <condition field="${sip_refer_to}">
+ <expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
+ <action application="set" data="refer_user=$1"/>
+ <action application="set" data="refer_domain=$2"/>
+ <action application="info"/>
+ <action application="bridge" data="sofia/${use_profile}/${refer_user}@${refer_domain}"/>
+ </condition>
+ </extension>
+ -->
+ <!--
+ This is an example of how to override the RURI on an outgoing invite to a registered contact.
+ -->
+ <!--
+ <extension name="ruri">
+ <condition field="destination_number" expression="^ruri$">
+ <action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|^[^\@]+(.*)|%1)}"/>
+ </condition>
+ </extension>
+
+ <extension name="7004">
+ <condition field="destination_number" expression="^\*7004$">
+ <action application="set" data="ruri_profile=default"/>
+ <action application="set" data="ruri_user=2000"/>
+ <action application="set" data="ruri_contact=1001@${domain_name}"/>
+ <action application="execute_extension" data="ruri"/>
+ </condition>
+ </extension>
+ -->
+
+ <!-- SEE WARNING ABOVE IF YOU ARE TRYING TO ADD EXTENSIONS HERE! -->
+
+ </context>
+</include>
diff --git a/config/freeswitch/dialplan.public.xml b/config/freeswitch/dialplan.public.xml
new file mode 100644
index 00000000..f30227e0
--- /dev/null
+++ b/config/freeswitch/dialplan.public.xml
@@ -0,0 +1,69 @@
+<!--
+ NOTICE:
+
+ This context is usually accessed via the external sip profile sitting on port 5080.
+
+ It is recommended to have separate inbound and outbound contexts. Not only for security
+ but clearing up why you would need to do such a thing. You don't want outside un-authenticated
+ callers hitting your default context which allows dialing calls thru your providers and results
+ in Toll Fraud.
+-->
+
+<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
+<include>
+ <context name="public">
+
+ <extension name="unloop">
+ <condition field="${unroll_loops}" expression="^true$"/>
+ <condition field="${sip_looped_call}" expression="^true$">
+ <action application="deflect" data="${destination_number}"/>
+ </condition>
+ </extension>
+ <!--
+ Tag anything pass thru here as an outside_call so you can make sure not
+ to create any routing loops based on the conditions that it came from
+ the outside of the switch.
+ -->
+ <extension name="outside_call" continue="true">
+ <condition>
+ <action application="set" data="outside_call=true"/>
+ </condition>
+ </extension>
+
+ <extension name="call_debug" continue="true">
+ <condition field="${call_debug}" expression="^true$" break="never">
+ <action application="info"/>
+ </condition>
+ </extension>
+
+ <!--
+ <extension name="public_extensions">
+ <condition field="destination_number" expression="^(10[01][0-9])$">
+ <action application="transfer" data="$1 XML default"/>
+ </condition>
+ </extension>
+ -->
+
+ <!--
+ You can place files in the public directory to get included.
+ -->
+ <X-PRE-PROCESS cmd="include" data="public/*.xml"/>
+ <!--
+ If you have made it this far lets challenge the caller and if they authenticate
+ lets try what they dialed in the default context. (commented out by default)
+ -->
+ <!--
+ <extension name="check_auth" continue="true">
+ <condition field="${sip_authorized}" expression="^true$" break="never">
+ <anti-action application="respond" data="407"/>
+ </condition>
+ </extension>
+
+ <extension name="transfer_to_default">
+ <condition>
+ <action application="transfer" data="${destination_number} XML default"/>
+ </condition>
+ </extension>
+ -->
+ </context>
+</include>
diff --git a/config/freeswitch/libncurses.so.5.7 b/config/freeswitch/libncurses.so.5.7
new file mode 100755
index 00000000..3b40374c
--- /dev/null
+++ b/config/freeswitch/libncurses.so.5.7
Binary files differ
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-23 12:42:42 +0000