Added following snippits about Asterisk

* A Basic Asterisk server for internal calling with IVR, curl triggers and message playback
* A Asterisk server to act as a bridge between a Bluetooth phone (as in/out bound route)
* A Node-Red Subflow to create a call between extensions using ARI

Linux/Config-Examples/Asterisk-basic-server/ReadMe.md

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+# Basic asterisk sever
+Go [Here](../../Docs/Setup-Asterisk-Server.md) to read about how to set it up
+
+
+**Config files that are empty**  
+these are just here to suppress errors
+* ```acl.conf```
+* ```ccss.conf```
+* ```cdr.conf```
+* ```cel.conf```
+* ```features.conf```
+* ```manager.conf``` 
+* ```pjproject.conf```
+* ```pjsip_wizard.conf```
+
+
+
+**Config files for basic usage**  
+If a config file does not have a comment, it does not require any tweaking  
+`*` Must be updated for each (new) extension  
+`**` Must be updated if (new) extension if a voip client needs to connect ({Soft}phone) or needs to be connected to (Homeassistant)
+* ```asterisk.conf```
+* ```extensions.conf``` Dial plan for extensions `*`
+* ```indications.conf``` Settings for dial tones
+* ```logger.conf```
+* ```modules.conf``` If not all functions are used you can disable some modules  
+* ```pjsip.conf``` Sets listening for PJSIP and loads sub configs
+* ```pjsip.extension.conf``` Defines the extension the credentials for the extensions `**`
+* ```pjsip.aor.conf``` Sets maximum connections for an extension `**`
+* ```pjsip.auth.conf``` Sets the credentials for the extensions`**`
+* ```udptl.conf```
+
+
+
+**Config files for use with ARI**  
+* ```ari.conf``` ARI basic config and credentials (Make sure to update this)
+* ```http.conf```  Settings for the integrated webserver for ARI
+* ```stasis.conf``` (Just here to suppress errors)

Linux/Config-Examples/Asterisk-basic-server/acl.conf

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+; Asterisk ACL configuration

Linux/Config-Examples/Asterisk-basic-server/ari.conf

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+; Asterisk ARI (Asterisk REST API) configuration
+[general]
+enabled = yes
+
+[aritest]
+type = user
+read_only = no
+password = testme
+password_format = plain

Linux/Config-Examples/Asterisk-basic-server/asterisk.conf

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+; Asterisk core configuration
+[directories](!)
+astcachedir => /tmp
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+astsbindir => /usr/sbin
+
+[options]
+;verbose = 3
+;debug = 3
+documentation_language = en_US

Linux/Config-Examples/Asterisk-basic-server/ccss.conf

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+; Asterisk ccss (Call Completion Supplementary Services) configuration

Linux/Config-Examples/Asterisk-basic-server/cdr.conf

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+; Asterisk cdr (Call Detail Record engine) configuration

Linux/Config-Examples/Asterisk-basic-server/cel.conf

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+; Asterisk CEL (Channel Event Logging) configuration

Linux/Config-Examples/Asterisk-basic-server/extensions.conf

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+; Asterisk exention configuration
+
+[from-internal]
+;Play audio message and hangup if 6010 is dialed
+exten = 6010,1,Answer()
+same = n,Wait(2)
+same = n,Playback(demo-congrats)
+same = n,Hangup()
+
+;Send http request, play audio message and hangup if 6011 is dialed
+exten = 6011,1,Answer()
+same = n,Wait(2)
+same = n,Set(CURL_RESULT=${CURL(http://<NoderedIP>:1880/endpoint/<ENDPOINT NAME>?exten=${EXTEN})})
+same = n,Playback(activated)
+same = n,Hangup()
+
+;Internal extentions (Phone,softphones or other Voip clients)
+; The number at the end (,30) sets the dial time out 
+exten => 6000,1,Dial(PJSIP/${EXTEN},30)
+
+;Connect to extenal the Homeassistent 'Assistent' service when 6020 is dialed
+exten => 6020,1,Dial(PJSIP/HomeAssistant,10)
+
+;IVR,
+;send to ivr if 6012 is dialed
+exten = 6012,1,Goto(ivr-test,s,1)
+
+;;The following section must be at the end of the file 
+[ivr-test]
+exten =>s,1,Answer()
+same =>n,set(IVRName=TestIVR)
+same => n,wait(2)
+same => n,Background(basic-pbx-ivr-main)
+same => n,WaitExten(10)
+;;Nr 1 to 4 are vaild options
+exten => _[1-4],1,Set(CURL_RESULT=${CURL(http://<NoderedIP>:1880/endpoint/<ENDPOINT NAME>?${IVRName}=${EXTEN})}
+same = n,Playback(activated)
+same = n,Playback(goodbye)
+same = n,Hangup()
+;;If option invalid
+exten => i,1,Playback(pm-invalid-option)
+exten => i,2,Goto(ivr-test,s,5)
+;; If timed out
+exten => t,1,Playback(vm-pls-try-again)
+exten => t,2,Goto(ivr-test,s,5)

Linux/Config-Examples/Asterisk-basic-server/features.conf

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+; Asterisk Call Features (transfer, monitor/mixmonitor, etc) configuration

Linux/Config-Examples/Asterisk-basic-server/http.conf

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+; Asterisk HTTP server (for ARI, etc) configuration
+[general]
+enabled=yes
+bindaddr=0.0.0.0
+bindport=8088

Linux/Config-Examples/Asterisk-basic-server/indications.conf

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+; Asterisk indication (location specific tone indications) configuration
+
+[general]
+country=nl
+
+[nl]
+description = Netherlands
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/500,0/9500
+dialrecall = 425/500,0/50
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425/500,0/50
+
+[us]
+description = United States / North America
+ringcadence = 2000,4000
+dial = 350+440
+busy = 480+620/500,0/500
+ring = 440+480/2000,0/4000
+congestion = 480+620/250,0/250
+callwaiting = 440/300,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

Linux/Config-Examples/Asterisk-basic-server/logger.conf

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+; Asterisk logging configuration
+
+[general]
+
+[logfiles]
+console => notice,warning,error
+messages => notice,warning,error

Linux/Config-Examples/Asterisk-basic-server/manager.conf

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+; Asterisk AMI - (Asterisk Manager Interface) configuration

Linux/Config-Examples/Asterisk-basic-server/modules.conf

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+; Asterisk Module Loader configuration file
+
+[modules]
+autoload=no
+
+;Core (BasicCalling)
+load = app_dial.so
+load = pbx_config.so
+load = pbx_loopback.so
+load = pbx_realtime.so
+load = pbx_spool.so
+load = func_callerid.so
+load = func_dialplan.so
+load = func_dialgroup.so
+load = res_ael_share.so
+load = res_timing_timerfd.so
+
+;Codecs (BasicCalling)
+load = codec_alaw.so
+load = codec_g722.so
+load = codec_g726.so
+load = codec_gsm.so
+load = codec_opus_open_source.so
+load = codec_resample.so
+load = codec_ulaw.so
+
+;Formats (BasicCalling)
+load = format_g719.so
+load = format_g726.so
+load = format_gsm.so
+load = format_ogg_speex.so
+load = format_siren14.so
+load = format_siren7.so
+load = format_wav.so
+load = format_wav_gsm.so
+load = res_format_attr_g729.so
+load = res_format_attr_opus.so
+load = res_format_attr_siren14.so
+load = res_format_attr_siren7.so
+
+;PJSIP (BasicCalling)
+require = chan_pjsip.so
+load = func_pjsip_aor.so
+load = func_pjsip_endpoint.so
+load = res_pjproject.so
+load = res_pjsip.so
+load = res_pjsip_authenticator_digest.so
+load = res_pjsip_caller_id.so
+load = res_pjsip_endpoint_identifier_ip.so
+load = res_pjsip_endpoint_identifier_user.so
+load = res_pjsip_exten_state.so
+load = res_pjsip_outbound_publish.so
+load = res_pjsip_outbound_registration.so
+load = res_pjsip_path.so
+load = res_pjsip_pubsub.so
+load = res_pjsip_refer.so
+load = res_pjsip_registrar.so
+load = res_pjsip_sdp_rtp.so
+load = res_pjsip_session.so
+
+;Sorcery (Rqired by PJSIP)
+load = res_sorcery_astdb.so
+load = res_sorcery_config.so
+load = res_sorcery_memory.so
+
+;RTP (Reqired by BasicCalling)
+load = chan_rtp.so
+load = bridge_native_rtp.so
+load = res_rtp_asterisk.so
+load = res_rtp_multicast.so
+
+;Bridge (Reqired by BasicCalling)
+load = bridge_simple.so
+load = bridge_builtin_features.so
+
+;FilePlayback (Reqired for playing a audio message for dialplan or IVR)
+load = app_controlplayback.so
+load = app_playback.so
+
+;;Make call from asterisk cli
+;load = app_originate.so
+;load = res_clioriginate.so 
+
+;ARI (Asterisk RESTful Interface)
+load => res_ari.so
+load => res_ari_applications.so
+load => res_ari_asterisk.so
+load => res_ari_bridges.so
+load => res_ari_channels.so
+load => res_ari_device_states.so
+load => res_ari_endpoints.so
+;load => res_ari_events.so
+load => res_ari_model.so
+load => res_ari_playbacks.so
+load => res_ari_recordings.so
+load => res_ari_sounds.so
+
+; stasis (Reqired by ARI)
+load => app_stasis.so
+load => res_stasis.so
+load => res_stasis_answer.so
+load => res_stasis_device_state.so
+load => res_stasis_playback.so
+load => res_stasis_recording.so
+load => res_stasis_snoop.so
+
+;cURL (Reqired for send curl commands form dialplan)
+load = func_curl.so
+load = res_curl.so

Linux/Config-Examples/Asterisk-basic-server/pjproject.conf

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+; Asterisk pjproject options

Linux/Config-Examples/Asterisk-basic-server/pjsip.aor.conf

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+; Asterisk AOR (address of record) config
+
+;Address for extenal the Homeassistent 'Assistent' service
+[HomeAssistant]
+type=aor
+qualify_frequency=0
+contact=sip:<HomeAssistant IP>:5060
+
+;Internal extentions (Phone,softphones or other Voip clients)
+[6000]
+type=aor
+
+; if one device is connected setting '1' is okay
+; when a device sometimes fails to register (like my Friz!box) try increasing it, '5' is a safe value 
+max_contacts=5

Linux/Config-Examples/Asterisk-basic-server/pjsip.auth.conf

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+; Asterisk PJSIP auth/credentials config
+
+;Credentials for extenal the Homeassistent 'Assistent' service
+[HomeAssistant]
+type=auth
+auth_type=userpass
+password=
+username=HA
+
+;Internal extentions (Phone,softphones or other Voip clients)
+[6000]
+type=auth
+auth_type=userpass
+password=<PASSWORD>
+username=6000

Linux/Config-Examples/Asterisk-basic-server/pjsip.conf

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+; Asterisk PJSIP config
+
+#include pjsip.extension.conf
+#include pjsip.auth.conf
+#include pjsip.aor.conf
+
+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0

Linux/Config-Examples/Asterisk-basic-server/pjsip.extension.conf

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+; Asterisk PJSIP  extention config
+
+;settings Homeassistent 'Assistent' service extention
+[HomeAssistant]
+type=endpoint
+context=from-external
+disallow=all
+allow=opus
+aors=HA
+send_connected_line=false
+rtp_keepalive=0
+user_eq_phone=no
+trust_id_inbound=no
+direct_media=no
+rtp_symmetric=yes
+auth=HomeAssistant
+aors=HomeAssistant
+;dtmf_mode=auto
+
+;Internal extentions (Phone,softphones or other Voip clients)
+
+[6000]
+type=endpoint
+context=from-internal
+callerid=DeskPhone <6000>
+disallow=all
+;Some phone may reqire other codecs,
+;the following are available ulaw,alaw,gsm,g726,g722,opus
+allow=g722
+auth=6000
+aors=6000
+

Linux/Config-Examples/Asterisk-basic-server/pjsip_wizard.conf

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+; Asterisk PJSIP Wizard configuration

Linux/Config-Examples/Asterisk-basic-server/stasis.conf

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+; Asterisk Stasis config
+[threadpool]
+
+[declined_message_types]

Linux/Config-Examples/Asterisk-basic-server/udptl.conf

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+; Asterisk udptl config
+
+[general]
+udptlstart=4000
+udptlend=4999
+udptlfecentries = 3
+udptlfecspan = 3
+use_even_ports = no