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prep release

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Jason D. McCormick 2025-06-08 15:36:05 -04:00
parent 1bc0aa2958
commit a6e99fcafb
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19 changed files with 30 additions and 1741 deletions
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13
asl3-init.sh
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@ -1,12 +1,23 @@
#!/usr/bin/bash
export DEBIAN_FRONTEND=noninteractive
# install the repo
apt update
apt upgrade -y
apt install -y wget
wget https://repo.allstarlink.org/public/asl-apt-repos.deb12_all.deb
dpkg -i asl-apt-repos.deb12_all.deb
apt update
apt install -y asl3-asterisk dahdi-dkms
# install kernel headers in preparation
apt install -y linux-headers-$(uname -r)
# prep /etc/asterisk
if [ ! -f /etc/asterisk/asterisk.conf ]; then
rm -rf /etc/asterisk/*
fi
chown asterisk:asterisk /etc/asterisk
# install
apt install -y asl3

12
asl3/asl3.sh
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@ -7,5 +7,17 @@ modprobe dahdi
modprobe dahdi_transcode
modprobe dahdi_dummy
# Prepare missing configuration (if needed)
chown asterisk:asterisk /etc/asterisk
if [ ! -f /etc/asterisk/asterisk.conf ]; then
apt install --reinstall -o Dpkg::Options::="--force-confmiss" asl3-asterisk-config
fi
# Fix defaults that don't work in a container
perl -pi -e 's/node_lookup_method\s+=\s+both/node_lookup_method = dns/g' \
/etc/asterisk/rpt.conf
perl -pi -e 's/^rxchannel\s+=\s+SimpleUSB\/1999/rxchannel = dahdi\/pseudo/g' \
/etc/asterisk/rpt.conf
/usr/sbin/asterisk -f

1
asl_backups/README Normal file
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@ -0,0 +1 @@
This is for backups created by asl-menu

4
asterisk_conf/README Normal file
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@ -0,0 +1,4 @@
All Asterisk configuration will be stored here. This
directory must be empty for a new composition of the
container as asl3-asterisk-config doesn't like
when existing content is present.

5
asterisk_conf/dnsmgr.conf
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@ -1,5 +0,0 @@
[general]
enable=yes ; enable creation of managed DNS lookups
; default is 'no'
refreshinterval=300 ; refresh managed DNS lookups every <n> seconds
; default is 300 (5 minutes)

56
asterisk_conf/echolink.conf
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@ -1,56 +0,0 @@
; If you are going to use this channel driver, you MUST enable it in modules.conf
; change:
; noload => chan_echolink.so ; echolink Channel Driver
; to:
; load => chan_echolink ; echolink Channel Driver
[el0]
call = INVALID ; Change this!
pwd = INVALID ; Change this!
name = YOUR NAME ; Change this!
qth = INVALID ; Change this!
email = INVALID ; Change this!
node = 000000 ; Change this!
; Data for EchoLink Status Page
lat = 0.0 ; Latitude in decimal degrees
lon = 0.0 ; Longitude in decimal degrees
freq = 0.0 ; not mandatory Frequency in MHz
tone = 0.0 ; not mandatory CTCSS Tone (0 for none)
power = 0 ; 0=0W, 1=1W, 2=4W, 3=9W, 4=16W, 5=25W, 6=36W, 7=49W, 8=64W, 9=81W (Power in Watts)
height = 0 ; 0=10 1=20 2=40 3=80 4=160 5=320 6=640 7=1280 8=2560 9=5120 (AMSL in Feet)
gain = 0 ; Gain in db (0-9)
dir = 0 ; 0=omni 1=45deg 2=90deg 3=135deg 4=180deg 5=225deg 6=270deg 7=315deg 8=360deg (Direction)
; Message to show on the echolink connect page. Use \n for new lines.
; message = This is the first line\nThis is another line\nThis is another line
maxstns = 20 ; Max Stations
rtcptimeout = 10 ; Max number of missed heartbeats from EL
recfile = /tmp/echolink_recorded.gsm ;
astnode = 1999 ; Change this!
context = radio-secure ; Default in code is echolink-in
; Max 4 servers
server1 = nasouth.echolink.org
server2 = naeast.echolink.org
server3 = servers.echolink.org
server4 = backup.echolink.org
; To deny w6xxx you would add the statement: deny = w6xxx
; To prohibit computer-based connections you would write: permit = *-*
; To allow access to only a select group of callsigns: permit = w6abc,w6def,...
; permit ; comma delimited list of callsign, type (-r)
; deny
; Remote text commands thru netcat:
; o.conip <IPaddress> (request a connect)
; o.dconip <IPaddress> (request a disconnect)
; o.rec (turn on/off recording)
; ipaddr
; port
#tryinclude custom/echolink.conf

175
asterisk_conf/extensions.conf
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@ -1,175 +0,0 @@
[general]
static = yes ; These two lines prevent the command-line interface
writeprotect = yes ; from overwriting the config file. Leave them here.
[globals]
HOMENPA = 999 ; change this to your Area Code
NODE = 1999 ; change this to your node number
[default]
exten => i,1,Hangup
[radio-secure]
;exten => ${NODE},1,rpt(${EXTEN})
exten => _XXXX!,1,NoOp(Connect from node: ${CALLERID(num)})
same => n,NoOp(Connect to: ${EXTEN})
;same => n,NoOp(The IAXPEER is ${IAXPEER(CURRENTCHANNEL)})
same => n,NoOp(The Channel IP is ${CHANNEL(peerip)})
;If channel IP eq localhost go to connect
same => n,GotoIf($["${CHANNEL(peerip)}" = "127.0.0.1"]?connect)
;If allowlist/extension doesn't exist check the denylist
same => n,GotoIf($[${DB_KEYCOUNT(allowlist/${EXTEN})} = 0]?denylist) ;goto denylist
;If allowlist/extension/callerid exists go to connect
same => n,GotoIf(${DB_EXISTS(allowlist/${EXTEN}/${CALLERID(num)})}?connect) ;goto connect
same => n,NoOp(${EXTEN} not in allowlist, Hangup)
same => n,Hangup
;If denylist/extension/callerID doesn't exist goto connect
same => n(denylist),GotoIf(${DB_EXISTS(denylist/${EXTEN}/${CALLERID(num)})}?:connect) ;don't goto connect
same => n,NoOp(${EXTEN} is in denylist, Hangup)
same => n,Hangup
same => n(connect),rpt(${EXTEN})
same => n,Hangup
[iaxrpt]
; Entered from iaxrpt in iax.conf
; Info: The X option passed to the Rpt application
; disables the normal security checks.
; Because incoming connections are validated in iax.conf,
; and we don't know where the user will be coming from in advance,
; the X option is required.
exten => ${NODE},1,rpt(${EXTEN}|X) ; NODE is the Name field in iaxrpt
[iax-client] ; for IAX VoIP clients.
exten => ${NODE},1,Ringing()
same => n,Wait(10)
same => n,Answer()
same => n,Set(CALLSIGN=${CALLERID(name)})
same => n,NoOp(Caller ID name is ${CALLSIGN})
same => n,NoOp(Caller ID number is ${CALLERID(number)})
same => n,GotoIf(${ISNULL(${CALLSIGN})}?hangit)
same => n,Playback(rpt/connected-to&rpt/node)
same => n,SayDigits(${NODE})
same => n,rpt(${NODE}|P|${CALLSIGN}-P)
same => n(hangit),NoOp(No Caller ID Name)
same => n,Playback(connection-failed)
same => n,Wait(1)
same => n,Hangup
; Comment-out the following clause if you want Allstar Autopatch service
[pstn-out]
exten => _NXXNXXXXXX,1,playback(ss-noservice)
same => n,Congestion
; Un-comment out the following clause if you want Allstar Autopatch service
;[pstn-out]
;exten => _NXXNXXXXXX,1,Dial(IAX2/allstar-autopatch/\${EXTEN})
; same => n,Busy
[invalidnum]
exten => s,1,Wait(3)
same => n,Playback(ss-noservice)
same => n,Wait(1)
same => n,Hangup
[radio]
exten => _X11,1,Goto(check_route,${EXTEN},1);
exten => _NXXXXXX,1,Goto(check_route,1${HOMENPA}${EXTEN},1)
exten => _1XXXXXXXXXX,1,Goto(check_route,${EXTEN},1)
exten => _07XX,1,Goto(parkedcalls,${EXTEN:1},1)
exten => 00,1,Goto(my-ip,s,1)
[check_route]
exten => _X.,1,Noop(${EXTEN})
; no 800
exten => _1800NXXXXXX,2,Goto(invalidnum,s,1)
exten => _1888NXXXXXX,2,Goto(invalidnum,s,1)
exten => _1877NXXXXXX,2,Goto(invalidnum,s,1)
exten => _1866NXXXXXX,2,Goto(invalidnum,s,1)
exten => _1855NXXXXXX,2,Goto(invalidnum,s,1)
; no X00 NPA
exten => _1X00XXXXXXX,2,Goto(invalidnum,s,1)
; no X11 NPA
exten => _1X11XXXXXXX,2,Goto(invalidnum,s,1)
; no X11
exten => _X11,2,Goto(invalidnum,s,1)
; no 555 Prefix in any NPA
exten => _1NXX555XXXX,2,Goto(invalidnum,s,1)
; no 976 Prefix in any NPA
exten => _1NXX976XXXX,2,Goto(invalidnum,s,1)
; no NPA=809
exten => _1809XXXXXXX,2,Goto(invalidnum,s,1)
; no NPA=900
exten => _1900XXXXXXX,2,Goto(invalidnum,s,1)
; okay, route it
exten => _1NXXXXXXXXX,2,Goto(pstn-out,${EXTEN:1},1)
exten => _X.,2,Goto(invalidnum,s,1)
[my-ip]
exten => s,1,Wait(1)
same => n,SayAlpha(${CURL(http://myip.vg)})
same => n,Hangup
[allstar-sys]
exten => _1.,1,Rpt(${EXTEN:1}|Rrpt/node:NODE:rpt/in-call:digits/0:PARKED|120)
exten => _1.,n,Hangup
exten => _2.,1,Ringing
exten => _2.,n,Wait(3)
exten => _2.,n,Answer
exten => _2.,n,Playback(rpt/node)
exten => _2.,n,Saydigits(${EXTEN:1})
exten => _2.,n,Rpt(${EXTEN:1}|P|${CALLERID(name)}-P)
exten => _2.,n,Hangup
exten => _3.,1,Ringing
exten => _3.,n,Wait(3)
exten => _3.,n,Answer
exten => _3.,n,Playback(rpt/node)
exten => _3.,n,Saydigits(${EXTEN:1})
exten => _3.,n,Rpt(${EXTEN:1}|Pv|${CALLERID(name)}-P)
exten => _3.,n,Hangup
exten => _4.,1,Ringing
exten => _4.,n,Wait(3)
exten => _4.,n,Answer
exten => _4.,n,Playback(rpt/node)
exten => _4.,n,Saydigits(${EXTEN:1})
exten => _4.,n,Rpt(${EXTEN:1}|D|${CALLERID(name)}-P)
exten => _4.,n,Hangup
exten => _5.,1,Ringing
exten => _5.,n,Wait(3)
exten => _5.,n,Answer
exten => _5.,n,Playback(rpt/node)
exten => _5.,n,Saydigits(${EXTEN:1})
exten => _5.,n,Rpt(${EXTEN:1}|Dv|${CALLERID(name)}-P)
exten => _5.,n,Hangup
[allstar-public]
exten => s,1,Ringing
same => n,Set(RESP=${CURL(https://register.allstarlink.org/cgi-bin/authwebphone.pl?${CALLERID(name)})})
same => n,Set(NODENUM=${CALLERID(number)})
same => n,GotoIf($["${RESP:0:1}" = "?"]?hangit)
same => n,GotoIf($["${RESP:0:1}" = ""]?hangit)
same => n,GotoIf($["${RESP:0:5}" != "OHYES"]?hangit)
same => n,Set(CALLSIGN=${RESP:5})
;exten => s,n,GotoIf($[${DB_EXISTS(whitelist/${CALLSIGN})}]?:hangit) ; Not in whitelist
same => n,Wait(3)
same => n,Playback(rpt/node,noanswer)
same => n,Saydigits(${NODENUM})
same => n,Set(CALLERID(name)=${CALLSIGN})
same => n,Set(CALLERID(num)=0)
same => n,Rpt(${NODENUM}|X)
same => n,Hangup
same => n(hangit),Answer
same => n,Wait(1)
same => n,Hangup
#tryinclude custom/extensions.conf

122
asterisk_conf/iax.conf
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@ -1,122 +0,0 @@
; Inter-Asterisk eXchange driver definition
[general]
; !!! IAX registration will be discontinued at some point !!!
; Setup rpt_http_registartions.conf instead.
; remove the leading ";"
;register => 1999:12345@register.allstarlink.org ; This must be changed to your node number, password
;register => 1998:12345@register.allstarlink.org
bindport = 4569 ; bindport and bindaddr may be specified
; NOTE: bindport must be specified BEFORE
; bindaddr or may be specified on a specific
; bindaddr if followed by colon and port
; (e.g. bindaddr=192.168.0.1:4569)
; bindaddr = 192.168.0.1 ; more than once to bind to multiple
; addresses, but the first will be the
; default
disallow = all ; The permitted codecs for outgoing connections
;Audio Quality Bandwidth
;allow = ulaw ; best 87 kbps
;allow = adpcm ; good 55 kbps
;allow = gsm ; mediocre 36 kbps
allow = ulaw
allow = adpcm
;allow = g722
;allow = g726aal2
allow = gsm
;allow = ilbc
jitterbuffer = yes
forcejitterbuffer = yes
dropcount = 2
maxjitterbuffer = 4000
maxjitterinterps = 10
resyncthreshold = 1000
maxexcessbuffer = 80
minexcessbuffer = 10
jittershrinkrate = 1
tos = 0x1E
autokill = yes
delayreject = yes
; iaxthreadcount = 30
; iaxmaxthreadcount = 150
; Incoming radio connections
[radio]
type = user
disallow = all
allow = ulaw
allow = adpcm
;allow = g722
;allow = g726aal2
allow = gsm
;allow = ilbc
codecpriority = host
context = radio-secure
transfer = no
[iaxrpt] ; Connect from iaxrpt Username field (PC AllStar Client)
type = user ; Notice type is user here <---------------
context = iaxrpt ; Context to jump to in extensions.conf
auth = md5
secret = Your_Secret_Pasword_Here
host = dynamic
disallow = all
allow = ulaw
allow = adpcm
allow = gsm
transfer = no
[iaxclient] ; Connect from iax client (Zoiper...)
type = friend ; Notice type here is friend <--------------
context = iax-client ; Context to jump to in extensions.conf
auth = md5
secret = Your_Secret_Password_Here
host = dynamic
disallow = all
allow = ulaw
allow = adpcm
allow = gsm
transfer = no
[allstar-sys]
type = user
context = allstar-sys
auth = rsa
inkeys = allstar
disallow = all
allow = ulaw
allow = adpcm
allow = gsm
[allstar-public]
type = user
context = allstar-public
auth = md5
secret = allstar
disallow = all
;calltokenoptional=0.0.0.0/0.0.0.0
requirecalltoken=no
allow = ulaw
allow = adpcm
allow = gsm
; The following should be un-commented to support Allstar Autopatch service
; [allstar-autopatch]
; type = peer
; host = register.allstarlink.org
; username = <One of the Node numbers on this server>
; secret = <The node password for the above node>
; auth = md5
; disallow = all
; allow = ulaw
; transfer = no
#tryinclude custom/iax.conf

20
asterisk_conf/logger.conf
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@ -1,20 +0,0 @@
;
; Logging Configuration
;
[general]
dateformat=%F %T.%3q ; with milliseconds
[logfiles]
;
; Format is "filename" and then "levels" of debugging to be included:
; debug
; notice
; warning
; error
;
; Special filename "console" represents the system console
;
;debug.log => debug
console => notice,warning,error,dtmf,debug,verbose
;console => notice,warning,error,dtmf
messages.log => notice,warning,error,dtmf

16
asterisk_conf/manager.conf
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@ -1,16 +0,0 @@
;
; Asterisk Call Management support
;
; By default asterisk will listen on localhost only.
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1 ; Comment when not localhost access
[admin]
secret = make4An1ce-secret
read = all,system,call,log,verbose,command,agent,user,config
write = all,system,call,log,verbose,command,agent,user,config
;deny = 0.0.0.0/0.0.0.0 ; Uncomment when not localhost
;permit=1.2.3.4/255.255.255.255 ; Uncomment when not localhost

186
asterisk_conf/modules.conf
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@ -1,186 +0,0 @@
;
; Asterisk module loading configuration file
;
;
; You can enable or disable any of the asterisk modules
;
; To enable a module, use : load = module_name.so
; To disable a module, use : noload = module_name.so
;
;
; For ASL (app_rpt), you need to enable the channel driver modules you will
; be using. You will find the modules in the "Channel Driver" section, below.
;
; The most common ASL (app_rpt) channel drivers are:
;
; chan_dahdi.so DAHDI Telephony
; chan_echolink.so Echolink Channel Driver
; chan_simpleusb.so Simple USB Radio Interface Channel Driver
; chan_usbradio.so USB Console Channel Driver
; chan_usrp.so USRP Channel Module
; chan_voter.so Voter Radio Channel Driver
;
[modules]
autoload = no
;load = func_frame_trace ; Debug
; Applications
load = app_authenticate.so ; Authentication Application
load = app_dial.so ; Dialing Application
load = app_exec.so ; Executes dialplan applications
noload = app_gps.so ; GPS Interface
load = app_playback.so ; Sound File Playback Application
require = app_rpt.so ; Radio Repeater/Remote Base Application
load = app_sendtext.so ; Send and Receive Text Applications
load = app_system.so ; Generic System() application
load = app_transfer.so ; Transfers a caller to another extension
; Channel Drivers
load = chan_dahdi.so ; DAHDI Telephony w/PRI & SS7 & MFC/R2
noload = chan_echolink.so ; Echolink Channel Driver
require = chan_iax2.so ; Inter Asterisk eXchange (Ver 2)
noload = chan_mobile.so ; Bluetooth Mobile Device Channel Driver
noload = chan_ooh323.so ; Objective Systems H323 Channel
load = chan_simpleusb.so ; SimpleUSB Radio Interface Channel Driver
noload = chan_tlb.so ; TheLinkBox Channel Driver
noload = chan_usbradio.so ; USB Console Channel Driver
noload = chan_usrp.so ; USRP Channel Module
noload = chan_voter.so ; Voter Radio Channel Driver
; Codecs
; CODEC AUDIO QUALITY BANDWIDTH (including IP and Ethernet headers)
; ULAW best 87 kilobits per second (kbps)
; ADPCM good 55 kbps
; GSM mediocre 36 kbps
; g726aal2
; ilbc
load = codec_adpcm.so ; Adaptive Differential PCM Coder/Decoder
load = codec_alaw.so ; A-law Coder/Decoder
load = codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder
noload = codec_dahdi.so ; Generic DAHDI Transcoder Codec Translator
noload = codec_g722.so ; ITU G.722-64kbps G722 Transcoder
load = codec_g726.so ; ITU G.726-32kbps G726 Transcoder
load = codec_gsm.so ; GSM Coder/Decoder
noload = codec_resample.so ; SLIN Resampling Codec
load = codec_ulaw.so ; mu-Law Coder/Decoder
; Formats
load = format_g723.so ; G.723.1 Simple Timestamp File Format
load = format_g726.so ; Raw G.726 (16/24/32/40kbps) data
noload = format_g729.so ; Raw G.729 data
load = format_gsm.so ; Raw GSM data
load = format_h263.so ; Raw H.263 data
load = format_h264.so ; Raw H.264 data
load = format_ilbc.so ; Raw iLBC data
load = format_mp3.so ; MP3 format [Any rate but 8000hz mono is optimal]
load = format_pcm.so ; Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz
load = format_sln.so ; Raw Signed Linear Audio support (SLN) 8khz-192khz
load = format_vox.so ; Dialogic VOX (ADPCM) File Format
load = format_wav_gsm.so ; Microsoft WAV format (Proprietary GSM)
load = format_wav.so ; Microsoft WAV/WAV16 format (8kHz/16kHz Signed Linear)
; Functions
load = func_base64.so ; base64 encode/decode dialplan functions
load = func_callerid.so ; Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)
load = func_cdr.so ; Call Detail Record (CDR) dialplan functions
load = func_channel.so ; Channel information dialplan functions
load = func_curl.so ; Load external URL
load = func_cut.so ; Cut out information from a string
load = func_db.so ; Database (astdb) related dialplan functions
load = func_enum.so ; ENUM related dialplan functions
load = func_env.so ; Environment/filesystem dialplan functions
load = func_global.so ; Variable dialplan functions
load = func_groupcount.so ; Channel group dialplan functions
load = func_logic.so ; Logical dialplan functions
load = func_math.so ; Mathematical dialplan function
load = func_md5.so ; MD5 digest dialplan functions
load = func_rand.so ; Random number dialplan function
load = func_realtime.so ; Read/Write/Store/Destroy values from a RealTime repository
noload = func_sha1.so ; SHA-1 computation dialplan function
load = func_strings.so ; String handling dialplan functions
noload = func_timeout.so ; Channel timeout dialplan functions
noload = func_uri.so ; URI encode/decode dialplan functions
; Core/PBX
load = pbx_config.so ; Text Extension Configuration
; Resources
load = res_crypto.so ; Cryptographic Digital Signatures
require = res_curl.so ; cURL Resource Module
require = res_rpt_http_registrations.so ; RPT HTTP Periodic Registrations
load = res_timing_dahdi.so ; DAHDI Timing Interface
load = res_timing_timerfd.so ; Timerfd Timing Interface is preferred for ASL3
require = res_usbradio.so ; USB Radio Resource
;
; The following modules are used by configurations that are setup for
; an "autopatch" or support SIP phones. If needed, uncomment each of
; the following lines (by removing the leading ";").
;
;load = bridge_builtin_features.so ; Built in bridging features
;load = bridge_builtin_interval_features.so ; Built in bridging interval features
;load = bridge_holding.so ; Holding bridge module
;load = bridge_native_rtp.so ; Native RTP bridging module
;load = bridge_simple.so ; Simple two channel bridging module
;load = bridge_softmix.so ; Multi-party software based channel mixing
;load = chan_bridge_media.so ; Bridge Media Channel Driver
;load = chan_pjsip.so ; PJSIP Channel Driver
;load = func_pjsip_endpoint.so ; Get information about a PJSIP endpoint
;load = func_sorcery.so ; Get a field from a sorcery object
;load = func_devstate.so ; Gets or sets a device state in the dialplan
;load = res_pjproject.so ; PJPROJECT Log and Utility Support
;load = res_pjsip_acl.so ; PJSIP ACL Resource
;load = res_pjsip_authenticator_digest.so ; PJSIP authentication resource
;load = res_pjsip_caller_id.so ; PJSIP Caller ID Support
;load = res_pjsip_dialog_info_body_generator.so ; PJSIP Extension State Dialog Info+XML Provider
;load = res_pjsip_diversion.so ; PJSIP Add Diversion Header Support
;load = res_pjsip_dtmf_info.so ; PJSIP DTMF INFO Support
;load = res_pjsip_endpoint_identifier_anonymous.so ; PJSIP Anonymous endpoint identifier
;load = res_pjsip_endpoint_identifier_ip.so ; PJSIP IP endpoint identifier
;load = res_pjsip_endpoint_identifier_user.so ; PJSIP username endpoint identifier
;load = res_pjsip_exten_state.so ; PJSIP Extension State Notifications
;load = res_pjsip_header_funcs.so ; PJSIP Header Functions
;load = res_pjsip_logger.so ; PJSIP Packet Logger
;load = res_pjsip_messaging.so ; PJSIP Messaging Support
;load = res_pjsip_mwi_body_generator.so ; PJSIP MWI resource
;load = res_pjsip_mwi.so ; PJSIP MWI resource
;load = res_pjsip_nat.so ; PJSIP NAT Support
;load = res_pjsip_notify.so ; CLI/AMI PJSIP NOTIFY Support
;load = res_pjsip_one_touch_record_info.so ; PJSIP INFO One Touch Recording Support
;load = res_pjsip_outbound_authenticator_digest.so ; PJSIP authentication resource
;load = res_pjsip_outbound_publish.so ; PJSIP Outbound Publish Support
;load = res_pjsip_outbound_registration.so ; PJSIP Outbound Registration Support
;load = res_pjsip_path.so ; PJSIP Path Header Support
;load = res_pjsip_pidf_body_generator.so ; PJSIP Extension State PIDF Provider
;load = res_pjsip_pidf_digium_body_supplement.so ; PJSIP PIDF Sangoma presence supplement
;load = res_pjsip_pidf_eyebeam_body_supplement.so ; PJSIP PIDF Eyebeam supplement
;load = res_pjsip_publish_asterisk.so ; PJSIP Asterisk Event PUBLISH Support
;load = res_pjsip_pubsub.so ; PJSIP event resource
;load = res_pjsip_refer.so ; PJSIP Blind and Attended Transfer Support
;load = res_pjsip_registrar.so ; PJSIP Registrar Support
;load = res_pjsip_rfc3326.so ; PJSIP RFC3326 Support
;load = res_pjsip_sdp_rtp.so ; PJSIP SDP RTP/AVP stream handler
;load = res_pjsip_send_to_voicemail.so ; PJSIP REFER Send to Voicemail Support
;load = res_pjsip_session.so ; PJSIP Session resource
;load = res_pjsip.so ; Basic SIP resource
;noload = res_pjsip_t38.so ; PJSIP T.38 UDPTL Support
;noload = res_pjsip_transport_websocket.so ; PJSIP WebSocket Transport Support
;load = res_pjsip_xpidf_body_generator.so ; PJSIP Extension State PIDF Provider
;load = res_rtp_asterisk.so ; Asterisk RTP Stack
;load = res_sorcery_astdb.so ; Sorcery Astdb Object Wizard
;load = res_sorcery_config.so ; Sorcery Configuration File Object Wizard
;load = res_sorcery_memory.so ; Sorcery In-Memory Object Wizard
;load = res_sorcery_realtime.so ; Sorcery Realtime Object Wizard

69
asterisk_conf/res_http_media_cache.conf
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;
; Sample configuration for res_http_media_cache
;
; res_http_media_cache is the HTTP backend for the core media cache. The
; following options can be used to tune the behavior of the implementation
; or left as default.
;
; See the module's and cURL's documentation for the exact meaning of these
; options.
[general]
; Maximum time in seconds the transfer is allowed to complete in.
;
; See https://curl.se/libcurl/c/CURLOPT_TIMEOUT.html for details.
;
;timeout_secs = 180
; The HTTP User-Agent to use for requests.
;
; See https://curl.se/libcurl/c/CURLOPT_USERAGENT.html for details.
;
;user_agent = asterisk-libcurl-agent/1.0
; Follow HTTP 3xx redirects on requests. This can be combined with the
; max_redirects option to limit the number of times a redirect will be
; followed per request.
;
; See https://curl.se/libcurl/c/CURLOPT_FOLLOWLOCATION.html for details.
;
;follow_location = false
; The maximum number of redirects to follow.
;
; See https://curl.se/libcurl/c/CURLOPT_MAXREDIRS.html for details.
;
;max_redirects = 8
; The HTTP/HTTPS proxy to use for requests. Leave unspecified to not use
; a proxy. This can be a URL with scheme, host and port.
;
; See https://curl.se/libcurl/c/CURLOPT_PROXY.html for details.
;
;proxy = https://localhost:1234
; The life-time for DNS cache entries.
;
; See https://curl.se/libcurl/c/CURLOPT_DNS_CACHE_TIMEOUT.html for details.
;
;dns_cache_timeout_secs = 60
; The comma separated list of allowed protocols for the request. Available with
; cURL version 7.85.0 or later.
; See https://curl.se/libcurl/c/CURLOPT_PROTOCOLS_STR.html for details.
;
;protocols = http,https
; The comma separated list of allowed protocols for redirects. Available with
; cURL version 7.85.0 or later. This can be used to prevent a redirect from
; a protocol like HTTPS to another supported protocol of cURL.
;
; See https://curl.se/libcurl/c/CURLOPT_REDIR_PROTOCOLS_STR.html for details.
;
;redirect_protocols = http,https

7
asterisk_conf/res_usbradio.conf
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[general]
;usb_devices = 1209:7388 ;comma delimited list of usb
;descriptors to allow.
;format vvvv:pppp in hexadecimal
;vvvv=vendor id, pppp=product id
;
;1209:7388 = AIOC (all in one cable)

662
asterisk_conf/rpt.conf
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@ -1,662 +0,0 @@
;*********************************************
;******** Template-tized rpt.conf ********
;*********************************************
; Note to editors: set tabs to 4 space characters. No wrap to keep comments neat.
; vim: tabstop=4
;
; Radio Repeater configuration file for use with app_rpt ASL3
;
[general]
;; Docker container only support dns
node_lookup_method = dns ;method used to lookup nodes
;both = dns lookup first, followed by external file (default)
;dns = dns lookup only
;file = external file lookup only
[nodes]
; If you are using automatic update for allstar link nodes, and you probably are,
; no allstar remote nodes should be defined here. Only place a definition
; for your local nodes, and private (not on allstar directory) nodes here.
; Sample nodes on this server. These are required. Use IP 127.0.0.1
; IAX port number if not the default 4569.
;1998 = radio@127.0.0.1/1998,NONE
; Sample public or private node behind your router.
;1001 = radio@192.168.1.8:4568/1001,NONE
; Sample private node on the internet.
;1000 = radio@8.8.8.8:4569/1000,NONE
1999 = radio@127.0.0.1/1999,NONE
[node-main](!)
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;; Template for all your nodes ;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; Set the defaults for your node(s) here.
; Add your nodes below the line that says
; Add you nodes below.
;;; rxchannel ;;;
; Must also be enabled in modules.conf
; Enable the selected channel driver in modules.conf !!!
; Rx/Tx audio/signalling channel. Choose ONLY 1 per node stanza.
; rxchannel = dahdi/pseudo ; No radio (hub)
; rxchannel = SimpleUSB/1999 ; SimpleUSB
; rxchannel = Radio/1999 ; USBRadio (DSP)
; rxchannel = Voter/1999 ; RTCM device
; rxchannel = Pi/1 ; Raspberry Pi PiTA
; rxchannel = Dahdi/1 ; PCI Quad card
; rxchannel = Beagle/1 ; BeagleBoard
; rxchannel = USRP/127.0.0.1:34001:32001 ;GNU Radio interface USRP
rxchannel = dahdi/pseudo ; No radio (hub)
duplex = 2 ; 0 = Half duplex with no telemetry tones or hang time.
; Special Case: Full duplex if linktolink is set to yes.
; This mode is preferred when interfacing with an external multiport repeater controller.
; Comment out idrecording and idtalkover to suppress IDs also
; 1 = Half duplex with telemetry tones and hang time. Does not repeat audio.
; This mode is preferred when interfacing a simplex node.
; 2 = Full Duplex with telemetry tones and hang time.
; This mode is preferred when interfacing a repeater.
; 3 = Full Duplex with telemetry tones and hang time, but no repeated audio.
; 4 = Full Duplex with telemetry tones and hang time. Repeated audio only when the autopatch is down.
linktolink = no ; disables forcing physical half-duplex operation of main repeater while
; still keeping half-duplex semantics (optional)
linkmongain = -22 ; Link Monitor Gain adjusts the audio level of monitored nodes when a signal from another node or the local receiver is received.
; If linkmongain is set to a negative number the monitored audio will decrease by the set amount in db.
; If linkmongain set to a positive number monitored audio will increase by the set amount in db.
; The value of linkmongain is in db. The default value is 0 db.
erxgain = -3 ; Echolink receive gain adjustment
; Note: Gain is in db-volts (20logVI/VO)
etxgain = 3 ; Echolink transmit gain adjustment
; Note: Gain is in db-volts (20logVI/VO)
;eannmode = 1 ; 1 = Say only node number on echolink connects (default = 1)
; 2 = say phonetic call sign only on echolink connects
; 3 = say phonetic call sign and node number on echolink connects
;echolinkdefault = 2 ; 0 = Telemetry output off
; 1 = Telemetry output on
; 2 = Timed telemetry output on command execution and for a short time thereafter
; 3 = Follow local telemetry mode
;echolinkdynamic = 1 ; 0 = Disallow users to change current echolink telemetry setting with a COP command
; 1 = Allow users to change the setting with a COP command
controlstates = controlstates ; system control state stanza
scheduler = schedule ; scheduler stanza
functions = functions ; Repeater Function stanza
phone_functions = functions ; Phone Function stanza
link_functions = functions ; Link Function stanza
telemetry = telemetry ; Telemetry stanza
morse = morse ; Morse stanza
wait_times = wait-times ; Wait times stanza
;inxlat = *,#,0123456789ABCD,Y ; The Y is for dialtone on function, a la CACTUS
; https://wiki.allstarlink.org/wiki/Rpt.conf#inxlat.3D
context = radio ; dialing context for phone
callerid = "Repeater" <0000000000> ; callerid for phone calls
accountcode = RADIO ; account code (optional)
hangtime = 2000 ; squelch tail hang time (in ms) (optional, default 5 seconds, 5000 ms)
althangtime = 4000 ; longer squelch tail
totime = 180000 ; transmit time-out time (in ms) (optional, default 3 minutes 180000 ms)
idrecording = |iNOTSET ; id recording or morse string
;idtalkover = ; Talkover ID (optional) default is none
idtime = 540000 ; id interval time (in ms) (optional) Default 5 minutes (300000 ms)
politeid = 30000 ; time in milliseconds before ID timer expires to try and ID in the tail. (optional, default 30000)
;tailmessagelist=/etc/asterisk/sounds/yourtailmessage
;tailsquashedtime=900000 ; 15 minutes
;tailmessagetime=21600000 ; 6 hours (200000000 ms, 55.5555 hours is the max value possible)
unlinkedct = ct2 ; Send a this courtesy tone when the user unkeys if the node is not connected to any other nodes. (optional, default is none)
remotect = ct3 ; remote linked courtesy tone (indicates a remote is in the list of links)
linkunkeyct = ct8 ; sent when a transmission received over the link unkeys
;nolocallinkct = 0 ; Send unlinkedct instead if another local node is connected to this node (hosted on the same PC).
;connpgm = yourconnectprogram ; Disabled. Execute a program you specify on connect. (default)
; passes 2 command line arguments to your program:
; 1. node number in this stanza (us)
; 2. node number being connected to us (them)
;discpgm = yourdisconnectprogram ; Disabled. Execute a program you specify on disconnect. (default)
; passes 2 command line arguments to your program:
; 1. node number in this stanza (us)
; 2. node number being disconnected from us (them)
;events=events ; Events Management https://wiki.allstarlink.org/wiki/Event_Management
;lnkactenable = 0 ; Set to 1 to enable the link activity timer. Applicable to standard nodes only.
;lnkacttime = 1800 ; Link activity timer time in seconds.
;lnkactmacro = *52 ; Function to execute when link activity timer expires.
;lnkacttimerwarn = 30seconds ; Message to play when the link activity timer has 30 seconds left.
;remote_inact_timeout = ; Specifies the amount of time without keying from the link. Set to 0 to disable timeout. (15 * 60)
;remote_timeout = ; Session time out for remote base. Set to 0 to disable. (60 * 60)
;remote_timeout_warning_freq = ; 30
;remote_timeout_warning = ; (3 * 60)
;nounkeyct = 0 ; Set to a 1 to eliminate courtesy tones and associated delays.
holdofftelem = 0 ; Hold off all telemetry when signal is present on receiver or from connected nodes
; except when an ID needs to be done and there is a signal coming from a connected node.
telemnomdb = -3 ; Telemetry Nominal Amplitude reference in dB
telemduckdb = -15 ; Telemetry Ducking in dB when local or link voice tx in progress
telemdefault = 2 ; 0 = telemetry output off
; 1 = telemetry output on (default = 1)
; 2 = timed telemetry output on command execution and for a short time thereafter.
telemdynamic = yes ; yes = disallow users to change the local telemetry setting with a COP command
; 1 = Allow users to change the setting with a COP command. (default = 1)
;beaconing = 0 ; Send ID regardless of repeater activity (Required in the UK, but probably illegal in the US)
parrotmode = 0 ; 0 = Parrot Off (default = 0)
; 1 = Parrot On Command
; 2 = Parrot Always
; 3 = Parrot Once by Command
parrottime = 1000 ; Set the amount of time in milliseconds
; to wait before parroting what was received
;rxnotch=1065,40 ; (Optional) Notch a particular frequency for a specified
; b/w. app_rpt must have been compiled with
; the notch option
;startup_macro = ; Best use in your node stanza (below) when more than one node
;nodenames = /var/lib/asterisk/sounds/rpt/nodenames.callsign ; Point to alternate nodename sound directory
;aprstt = general ; To enable APRStt, set this value to the stanza to use in gps.conf
; module app_gps.so will need to be enabled
; *** Status Reporting ***
;
; Uncomment the following "statpost_url" line to enable status reporting
; of all of your nodes to "stats.allstarlink.org". To enable reporting
; of a single node then add the line (without the leading ";") to the
; per-node stanza(s).
;
;statpost_url = http://stats.allstarlink.org/uhandler ; Status updates
; *** Audio Archiving ***
;
; The following "archivedir" line can be used to enable a simple log and
; audio recorder of the activity on a node. When enabled, a series of
; recordings, one for each active COR on the node. The file(s) will be
; named with the date and time down to the 1/100th of a second. This
; logging can be useful in debugging, policing, or other creative things.
;
; The "archiveformat" line can be used to specify the format of the audio
; recordings. By default, the format will be "wav49" (GSM in a .WAV file).
; Other options you may consider include "wav" (SLIN in a .wav file) and
; "gsm" (GSM in straight gsm format).
;
; The "archivedir" and "archiveformat" lines can be enabled here (affecting
; all nodes) or in the per-node stanzas (for recording of individual nodes).
;
; Note: enabling these recordings can adversly impact the CPU utilization
; on the device and consume large amounts of the available storage.
;
;archivedir = /var/spool/asterisk/monitor ; top-level recording directory
;archiveformat = wav49 ; audio format (default = wav49)
;;; End of node-main template
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;; Configure your nodes here ;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
; Note: there is no need to duplicate entire settings. Only
; place settings that are different than the template.
;
[1999](node-main)
;;;;;;;;;;;;;;;;;;; Your node settings here ;;;;;;;;;;;;;;;;;;;
rxchannel = dahdi/pseudo ; pseudo
;startup_macro = *8132000
;[1998](node-main)
;;;;;;;;;;;;;;;;;;; Another node settings here ;;;;;;;;;;;;;;;;;;;
;startup_macro = *8132000
;morse = morse_1998 ; Sample morse stanza for node 1998
[functions]
;;;;;;;;;;;;;;;;;;; functions stanza ;;;;;;;;;;;;;;;;;;;
; Prefix Functions
; *1 Disconnect Link
; *2 Monitor Link
; *3 Connect Link
; *4 Command Mode
; *5 Macros
; *6 User Functions
; *7 Connection Status/Functions
; *8 User Functions
; *9 User Functions
; *0 User Functions
; *A User Functions
; *B User Functions
; *C User Functions
; *D User Functions
;;;;;; Mandatory Command Codes ;;;;;
1 = ilink,1 ; Disconnect specified link
2 = ilink,2 ; Connect specified link -- monitor only
3 = ilink,3 ; Connect specified link -- tranceive
4 = ilink,4 ; Enter command mode on specified link
70 = ilink,5 ; System status
99 = cop,6 ; PTT (phone mode only)
;;;;; Macro Commands ;;;;;
5 = macro
;;;;; Autopatch Commands ;;;;;
; Note, This may be a good place for other 2 digit frequently used commands
;61 = autopatchup,noct = 1,farenddisconnect = 1,dialtime = 20000 ; Autopatch up
;62 = autopatchdn ; Autopatch down
; autopatchup can optionally take comma delimited setting=value pairs:
; context = string ; Override default context with "string"
; dialtime = ms ; Specify the max number of milliseconds between phone number digits (1000 milliseconds = 1 second)
; farenddisconnect = 1 ; Automatically disconnect when called party hangs up
; noct = 1 ; Don't send repeater courtesy tone during autopatch calls
; quiet = 1 ; Don't send dial tone, or connect messages. Do not send patch down message when called party hangs up
; Example: 123=autopatchup,dialtime=20000,noct=1,farenddisconnect=1
;;;;; Status Commands ;;;;;
; 1 - Force ID (global)
; 2 - Give Time of Day (global)
; 3 - Give software Version (global)
; 4 - Give GPS location info
; 5 - Last (dtmf) user
; 11 - Force ID (local only)
; 12 - Give Time of Day (local only)
721 = status,1 ; Force ID (global)
722 = status,2 ; Give Time of Day (global)
723 = status,3 ; Give software Version (global)
724 = status,4 ; Give GPS location info
725 = status,5 ; Last (dtmf) user
711 = status,11 ; Force ID (local only)
712 = status,12 ; Give Time of Day (local only)
;;;;; Link Commands ;;;;;
; 1 - Disconnect specified link
; 2 - Connect specified link -- monitor only
; 3 - Connect specified link -- tranceive
; 4 - Enter command mode on specified link
; 5 - System status
; 6 - Disconnect all links
; 7 - Last Node to Key Up
; 8 - Connect specified link -- local monitor only
; 9 - Send Text Message (9,<destnodeno or 0 (for all)>,Message Text, etc.
; 10 - Disconnect all RANGER links (except permalinks)
; 11 - Disconnect a previously permanently connected link
; 12 - Permanently connect specified link -- monitor only
; 13 - Permanently connect specified link -- tranceive
; 15 - Full system status (all nodes)
; 16 - Reconnect links disconnected with "disconnect all links"
; 17 - MDC test (for diag purposes)
; 18 - Permanently Connect specified link -- local monitor only
;;;;; ilink commands ;;;;;
; commands 1 through 5 are defined in the Mandatory Command section
; 806 = ilink,6 ; Disconnect all links
; 807 = ilink,7 ; Last Node to Key Up
; 808 = ilink,8 ; Connect specified link -- local monitor only
; 809 = ilink,9,1999,"Testing" ; would send a text message to node 1999 replace 1999 with 0 for all connected nodes
; 810 = ilink,10 ; Disconnect all RANGER links (except permalinks)
; 811 = ilink,11 ; Disconnect a previously permanently connected link
; 812 = ilink,12 ; Permanently connect specified link -- monitor only
; 813 = ilink,13 ; Permanently connect specified link -- tranceive
; 815 = ilink,15 ; Full system status (all nodes)
; 816 = ilink,16 ; Reconnect links disconnected with "disconnect all links"
; 817 = ilink,17 ; MDC test (for diag purposes)
; 818 = ilink,18 ; Permanently Connect specified link -- local monitor only
;;;;; Control operator (cop) functions. ;;;;;
;Change these to something other than these codes listed below!
;901 = cop,1 ; System warm boot
;902 = cop,2 ; System enable
;903 = cop,3 ; System disable
;904 = cop,4 ; Test tone on/off (toggle)
;905 = cop,5 ; Dump system variables on console (debug use only)
;907 = cop,7 ; Time out timer enable
;908 = cop,8 ; Time out timer disable
;909 = cop,9 ; Autopatch enable
;910 = cop,10 ; Autopatch disable
;911 = cop,11 ; User linking functions enable
;912 = cop,12 ; User linking functions disable
;913 = cop,13 ; Query system control state
;914 = cop,14 ; Set system control state
;915 = cop,15 ; Scheduler enable
;916 = cop,16 ; Scheduler disable
;917 = cop,17 ; User functions enable (time, id, etc)
;918 = cop,18 ; User functions disable
;919 = cop,19 ; Select alternate hang time (althangtime)
;920 = cop,20 ; Select standard hangtime (hangtime)
;921 = cop,21 ; Enable Parrot Mode
;922 = cop,22 ; Disable Parrot Mode
;923 = cop,23 ; Birdbath (Current Parrot Cleanup/Flush)
;924 = cop,24 ; Flush all telemetry
;925 = cop,25 ; Query last node un-keyed
;926 = cop,26 ; Query all nodes keyed/unkeyed
;930 = cop,30 ; Recall Memory Setting in Attached Xcvr
;931 = cop,31 ; Channel Selector for Parallel Programmed Xcvr
;932 = cop,32 ; Touchtone pad test: command + Digit string + # to playback all digits pressed
;933 = cop,33 ; Local Telemetry Output Enable
;934 = cop,34 ; Local Telemetry Output Disable
;935 = cop,35 ; Local Telemetry Output on Demand
;936 = cop,36 ; Foreign Link Local Output Path Enable
;937 = cop,37 ; Foreign Link Local Output Path Disable
;938 = cop,38 ; Foreign Link Local Output Path Follows Local Telemetry
;939 = cop,39 ; Foreign Link Local Output Path on Demand
;942 = cop,42 ; Echolink announce node # only
;943 = cop,43 ; Echolink announce node Callsign only
;944 = cop,44 ; Echolink announce node # & Callsign
;945 = cop,45 ; Link Activity timer enable
;945 = cop,46 ; Link Activity timer disable
;947 = cop,47 ; Reset "Link Config Changed" Flag
;948 = cop,48 ; Send Page Tone (Tone specs separated by parenthesis)
; 949 = cop,49 ; Disable incoming connections (control state noice)
; 950 = cop,50 ; Enable incoming connections (control state noicd)
; 951 = cop,51 ; Enable sleep mode
; 952 = cop,52 ; Disable sleep mode
; 953 = cop,53 ; Wake up from sleep
; 954 = cop,54 ; Go to sleep
; 955 = cop,55 ; Parrot Once if parrot mode is disabled
; 956 = cop,56 ; Rx CTCSS Enable
; 957 = cop,57 ; Rx CTCSS Disable
; 958 = cop.58 ; Tx CTCSS On Input only Enable
; 959 = cop,59 ; Tx CTCSS On Input only Disable
; 960 = cop,60 ; Send MDC-1200 Burst (cop,60,type,UnitID[,DestID,SubCode])
; Type is 'I' for PttID, 'E' for Emergency, and 'C' for Call
; (SelCall or Alert), or 'SX' for STS (ststus), where X is 0-F.
; DestID and subcode are only specified for the 'C' type message.
; UnitID is the local systems UnitID. DestID is the MDC1200 ID of
; the radio being called, and the subcodes are as follows:
; Subcode '8205' is Voice Selective Call for Spectra ('Call')
; Subcode '8015' is Voice Selective Call for Maxtrac ('SC') or
; Astro-Saber('Call')
; Subcode '810D' is Call Alert (like Maxtrac 'CA')
; 961 = cop,61 ; Send Message to USB to control GPIO pins (cop,61,GPIO1=0[,GPIO4=1].....)
; 962 = cop,62 ; Send Message to USB to control GPIO pins, quietly (cop,62,GPIO1=0[,GPIO4=1].....)
; 963 = cop,63 ; Send pre-configred APRStt notification (cop,63,CALL[,OVERLAYCHR])
; 964 = cop,64 ; Send pre-configred APRStt notification, quietly (cop,64,CALL[,OVERLAYCHR])
; 965 = cop,65 ; Send POCSAG page (equipped channel types only)
; [functions-remote]
;;;;; Functions for remote bases ;;;;;
;
; 0 = remote,1 ; Retrieve Memory
; 1 = remote,2 ; Set freq.
; 2 = remote,3 ; Set tx PL tone
; 3 = remote,4 ; Set rx PL tone
; 40 = remote,100 ; Rx PL off
; 41 = remote,101 ; Rx PL on
; 42 = remote,102 ; Tx PL off
; 43 = remote,103 ; Tx PL on
; 44 = remote,104 ; Low Power
; 45 = remote,105 ; Medium Power
; 46 = remote,106 ; High Power
; 711 = remote,107 ; Bump -20
; 714 = remote,108 ; Bump -100
; 717 = remote,109 ; Bump -500
; 713 = remote,110 ; Bump +20
; 716 = remote,111 ; Bump +100
; 719 = remote,112 ; Bump +500
; 721 = remote,113 ; Scan - slow
; 724 = remote,114 ; Scan - quick
; 727 = remote,115 ; Scan - fast
; 723 = remote,116 ; Scan + slow
; 726 = remote,117 ; Scan + quick
; 729 = remote,118 ; Scan + fast
; 79 = remote,119 ; Tune
; 51 = remote,5 ; Long status query
; 52 = remote,140 ; Short status query
; 67 = remote,210 ; Send a *
; 69 = remote,211 ; Send a #
; 91 = remote,99,CALLSIGN,LICENSETAG ; Remote base login.
; Define a different dtmf sequence for each user which is
; authorized to use the remote base to control access to it.
; For examble 9139583=remote,99,WB6NIL,G would grant access to
; the remote base and announce WB6NIL as being logged in.
; Another entry, 9148351=remote,99,WA6ZFT,E would grant access to
; the remote base and announce WA6ZFT as being logged in.
; When the remote base is disconnected from the originating node, the
; user will be logged out. The LICENSETAG argument is used to enforce
; tx frequency limits. See [txlimits] below.
; 85 = cop,6 ; Remote base telephone key
[telemetry]
;;;;; Telemetry ;;;;;
; Telemetry entries can be shared across all repeaters, or defined for each repeater.
; Can be a tone sequence, morse string, or a file
;
; |t - Tone escape sequence
;
; Tone sequences consist of 1 or more 4-tuple entries (freq1, freq2, duration, amplitude)
; Single frequencies are created by setting freq1 or freq2 to zero.
;
; |m - Morse escape sequence
;
; Sends Morse code at the telemetry amplitude and telemetry frequency as defined in the
; [morse] section.
;
; Follow with an alphanumeric string
;
; |i - Morse ID escape sequence
;
; Sends Morse code at the ID amplitude and ID frequency as defined in the
; [morse] section.
;
ct1 = |t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
ct2 = |t(660,880,150,2048)
ct3 = |t(440,0,150,4096)
ct4 = |t(550,0,150,2048)
ct5 = |t(660,0,150,2048)
ct6 = |t(880,0,150,2048)
ct7 = |t(660,440,150,2048)
ct8 = |t(700,1100,150,2048)
ct9 = |t(1633,0,50,1000)(0,0,30,0)(1209,0,50,1000);
ranger = |t(1800,0,60,3072)(0,0,50,0)(1800,0,60,3072)(0,0,50,0)(1800,0,60,3072)(0,0,50,0)(1800,0,60,3072)(0,0,50,0)(1800,0,60,3072)(0,0,50,0)(1800,0,60,3072)(0,0,150,0)
remotemon = |t(1209,0,50,2048) ; local courtesy tone when receive only
remotetx = |t(1633,0,50,3000)(0,0,80,0)(1209,0,50,3000) ; local courtesy tone when linked Trancieve mode
cmdmode = |t(900,903,200,2048)
functcomplete = |t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
remcomplete = |t(650,0,100,2048)(0,0,100,0)(650,0,100,2048)(0,0,100,0)(650,0,100,2048)
pfxtone = |t(350,440,30000,3072)
patchup = rpt/callproceeding
patchdown = rpt/callterminated
; As far as what the numbers mean,
; (000,000,010,000)
; | | | |-------amplitude
; | | |-------------duration
; | |-------------------Tone 2
; |-------------------------Tone 1
; So, with 0,0,10,0 That says No Tone1, No Tone2, 10ms duration, 0 Amplitude.
; Use it for a delay. Fine tuning for how long before telemetry is sent, in conjunction with the telemdelay parameter)
; The numbers, like 350,440,10,2048 are 350Hz, 440Hz, 10ms delay, amplitude of 2048.
[morse]
;;;;; Mores code ;;;;;
; Morse code parameters, these are common to all repeaters.
speed = 20 ; Approximate speed in WPM
frequency = 800 ; Morse Telemetry Frequency
amplitude = 4096 ; Morse Telemetry Amplitude
idfrequency = 1065 ; Morse ID Frequency
idamplitude = 1024 ; Morse ID Amplitude
;[morse]
;speed = 35 ; Not leagle in the USA
;frequency = 800 ; Morse Telemetry Frequency *Changed
;amplitude = 4096 ; Morse Telemetry Amplitude
;idfrequency = 750 ; Morse ID Frequency *Changed
;idamplitude = 512 ; Morse ID Amplitude *Changed
[wait-times]
; This section allows wait times for telemetry events to be adjusted
; A section for wait times can be defined for every repeater
telemwait = 2000 ; Time to wait before sending most telemetry
idwait = 500 ; Time to wait before starting ID
unkeywait = 2000 ; Time to wait after unkey before sending CT's and link telemetry
calltermwait = 2000 ; Time to wait before announcing "call terminated"
[wait-times_hd]
; Sample for half-duplex
telemwait = 100 ; Time to wait before sending most telemetry
idwait = 100 ; Time to wait before starting ID
unkeywait = 100 ; Time to wait after unkey before sending CT's and link telemetry
calltermwait = 2000 ; Time to wait before announcing "call terminated"
[memory]
;;;;; Remote base memories ;;;;;
; Memories for remote bases (not for repeaters or hotspots). Seldom used.
;00 = 146.580,100.0,m
;01 = 147.030,103.5,m+t
;02 = 147.240,103.5,m+t
;03 = 147.765,79.7,m-t
;04 = 146.460,100.0,m
;05 = 146.550,100.0,m
[macro]
;;;;; Macro commands ;;;;;
;1 = *812000
;2 = *822000
;3 = *832000
;4 = *31998*32000
;[daq-list]
;;;;; Data Acquisition configuration ;;;;;
;Where: device_name1 and device_name2 are stanzas you define in this file
;device = daq-cham-1
;device = device_name1
;device = device_name2
;[daq-cham-1] ; Defined in [daq-list]
; Device name
;hwtype = uchameleon ; DAQ hardware type
;devnode = /dev/ttyUSB0 ; DAQ device node (if required)
;1 = inadc ; Pin definition for an ADC channel
;2 = inadc
;3 = inadc
;4 = inadc
;5 = inadc
;6 = inadc
;7 = inadc
;8 = inadc
;9 = inp ; Pin definition for an input with a weak pullup resistor
;10 = inp
;11 = inp
;12 = inp
;13 = in ; Pin definition for an input without a weak pullup resistor
;14 = out ; Pin definition for an output
;15 = out
;16 = out
;17 = out
;18 = out
;[meter-faces]
;
;face = scale(scalepre,scalediv,scalepost),word/?,...
;
; scalepre = offset to add before dividing with scalediv
; scalediv = full scale/number of whole units (e.g. 256/20 or 12.8 for 20 volts).
; scalepost = offset to add after dividing with scalediv
;
;face = range(X-Y:word,X2-Y2:word,...),word/?,...
;face = bit(low-word,high-word),word/?,...
;
; word/? is either a word in /var/lib/asterisk/sounds or one of its subdirectories,
; or a question mark which is a placeholder for the measured value.
;
;
; Battery voltage 0-20 volts
;batvolts = scale(0,12.8,0),rpt/thevoltageis,?,ha/volts
; 4 quadrant wind direction
;winddir = range(0-33:north,34-96:west,97-160:south,161-224:east,225-255:north),rpt/thewindis,?
; LM34 temperature sensor with 130 deg. F full scale
;lm34f = scale(0,1.969,0),rpt/thetemperatureis,?,degrees,fahrenheit
; Status poll (non alarmed)
;light = bit(ha/off,ha/on),ha/light,?
;[alarms]
;
;tag = device,pin,node,ignorefirst,func-low,func-hi
;
;tag = a unique name for the alarm
;device = daq device to poll
;pin = the device pin to be monitored
;ignorefirstalarm = set to 1 to throwaway first alarm event, or 0 to report it
;node = the node number to execute the function on
;func-low = the DTMF function to execute on a high to low transition
;func-high = the DTMF function to execute on a low to high transition
;
; a '-' as a function name is shorthand for no-operation
;
;door = daq-cham-1,9,1,2017,*7,-
;pwrfail = daq-cham-1,10,0,2017,*911111,-
;[events]
;;;;; Events Management ;;;;;
;status,2 = c|f|RPT_NUMLINKS ; Say time of day when all links disconnect.
[controlstates]
;;;;; Control states ;;;;;
; Allow several control operator functions to be changed at once using one command (good for scheduling)
;statenum = copcmd,[copcmd]... https://wiki.allstarlink.org/wiki/Rpt.conf#Control_States_Stanza
0 = rptena,lnkena,apena,totena,ufena,noicd ; Normal operation
1 = rptena,lnkena,apdis,totdis,ufena,noice ; Net and news operation
2 = rptena,lnkdis,apdis,totena,ufdis,noice ; Repeater only operation
[schedule]
;;;;; Scheduler - execute a macro at a given time ;;;;;
;dtmf_function = m h dom mon dow ; ala cron, star is implied
;2 = 00 00 * * * ; at midnight, execute macro 2.

5
asterisk_conf/rpt_http_registrations.conf
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@ -1,5 +0,0 @@
[general]
register_interval = 180 ; every 3 minutes
[registrations]
register => 1234:abcdef@register.allstarlink.org

145
asterisk_conf/simpleusb.conf
View file

@ -1,145 +0,0 @@
;***************************************************
;******** Template-tized simpleusb.conf ********
;***************************************************
; Note to editors: set tabs to 4 space characters. No wrap to keep comments neat.
; vim: tabstop=4
;
; SimpleUSB channel driver Configuration File
;
;;;;; New to ASL3 ;;;;;
;;;;; The SimpleUSB "tune" settings have moved to ;;;;;
;;;;; this file. The simpleusb_tune_usb_1999.conf ;;;;;
;;;;; file is no longer used. ;;;;;
; If you are going to use this channel driver, you MUST enable it in modules.conf
; change:
; noload => chan_simpleusb.so ; CM1xx USB Cards with Radio Interface Channel Driver (No DSP)
; noload => res_usbradio.so ; Required for both simpleusb and usbradio
; to:
; load => chan_simpleusb.so ; CM1xx USB Cards with Radio Interface Channel Driver (No DSP)
; load => res_usbradio.so ; Required for both simpleusb and usbradio
[general]
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; simpleusb channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The simpleusb channel can't accept jitter,
; thus an enabled jitterbuffer on the receive simpleusb side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an simpleusb
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
[node-main](!)
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;; Template for all your SimpleUSB nodes ;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
eeprom = 0 ; EEPROM installed: 0,1
; 0 = no (default)
; 1 = yes
hdwtype = 0 ; Leave this set to 0 for USB sound fobs designed for app_rpt
; or modified cmxxx sound fobs.
; 1 is for Dingotel/Sph interfaces.
; 2 is for NHC interfaces.
; 3 is for a custom interface.
; Receiver parameters
rxboost = no ; no = 20db attenuator inserted, yes = 20db attenuator removed
; Set to yes for additional gain if using a low-level receiver output
; CM119B devices are hard coded to rxboost 20dB attenuator removed
carrierfrom = usbinvert ; no,usb,usbinvert
; no - no carrier detection at all
; usb - from the COR line on the USB sound fob (Active high)
; usbinvert - from the inverted COR line on the USB sound fob (Active low)
; pp - from parallel port (Active high)
; ppinvert - from the inverted parallel port (Active low)
ctcssfrom = usbinvert ; no,usb,usbinvert
; no - CTCSS decoding, system will be carrier squelch
; usb - CTCSS decoding using input from USB sound fob (Active high)
; usbinvert - from the inverted CTCSS line on the USB sound fob (Active low)
; pp - from parallel port (Active high)
; ppinvert - from the inverted parallel port (Active low)
deemphasis = no ; enable de-emphasis (input from discriminator)
plfilter = yes ; enable PL filter
rxondelay = 0 ; Number of 20mSec intervals following the release of PTT.
; Uncomment and/or adjust for simplex nodes to eliminate "Ping Ponging"
; or "Relay Racing". A positive value here will instruct the usbradio
; driver to ignore the COR line for a specified number of 20mSec
; intervals following the release of PTT. Use this ONLY on simplex
; nodes, and leave commented out for repeaters or other full duplex nodes.
txoffdelay = 0 ; Ignore the receiver for a specified number of 20 millisecond
; intervals after the transmitter unkeys.
; This is useful when setting up a half-duplex link with an existing
; repeater, where you need to ignore the repeater's hangtime.
; Transmitter parameters
txmixa = voice ; Left channel output (A): no,voice
; no - Do not output anything
; voice - output voice only
txmixb = no ; Right channel output (B): no,voice
; no - Do not output anything
; voice - output voice only
invertptt = no ; Invert PTT: no,yes
; no - ground to transmit
; yes - open to transmit
preemphasis = no ; Perform standard 6db/octave pre-emphasis
; pager = no ; no,a,b (e.g. pager = b means "put the normal repeat audio on channel A, and the pager audio on channel B")
; duplex3 = 0 ; duplex 3 gain setting (0 to disable)
clipledgpio = 1 ; Enable ADC Clip Detect feature to use a GPIO output (0 to disable).
; Supports URI Clip LED by setting a GPIO (if available) high for 500mS
; when clipping detected. Value = GPIO# to use (GPIO1 recommended)
legacyaudioscaling = no ; If yes, continue to do raw audio sample scaling and clipping, resulting in Tx audio levels increasing
; by 0.78dB and Rx audio levels increasing by 0-1.5dB. Should be set to no unless you have an existing
; node with precisely adjusted audio levels and are unable to adjust them. This parameter and associated
; scaling/clipping code will be deleted once existing installs have been able to verify their audio levels
;;; End of node-main template
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;; Configure your nodes here ;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
; Note: there is no need to duplicate entire settings. Only
; place settings that are different than the template.
;
; Note: the device string is automatically found when the
; USB setting "devstr=" is empty.
;
; Note: the interface "tune" settings will be added to the
; per-node settings (below).
[1999](node-main)
;;;;; ASL3 Tune settings ;;;;;
devstr=
rxmixerset=500
txmixaset=500
txmixbset=500

213
asterisk_conf/usbradio.conf
View file

@ -1,213 +0,0 @@
;**************************************************
;******** Template-tized usbradio.conf ********
;**************************************************
; Note to editors: set tabs to 4 space characters. No wrap to keep comments neat.
; vim: tabstop=4
;
; USBRadio channel driver Configuration File
;
;;;;; New to ASL3 ;;;;;
;;;;; The USBRadio "tune" settings have moved to ;;;;;
;;;;; this file. The usbradio_tune_usb_1999.conf ;;;;;
;;;;; file is no longer used. ;;;;;
; If you are going to use this channel driver, you MUST enable it in modules.conf
; noload => chan_usbradio.so ; CM1xx USB Cards with Radio Interface Channel Driver (DSP)
; noload => res_usbradio.so ; Required for both simpleusb and usbradio
; to:
; load => chan_usbradio.so ; CM1xx USB Cards with Radio Interface Channel Driver (DSP)
; load => res_usbradio.so ; Required for both simpleusb and usbradio
[general]
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; USBRADIO channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The USBRADIO channel can't accept jitter,
; thus an enabled jitterbuffer on the receive USBRADIO side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an USBRADIO
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
[node-main](!)
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;; Template for all your USBRadio nodes ;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
eeprom = 0 ; EEPROM installed: 0,1
; 0 = no (default)
; 1 = yes
hdwtype = 0 ; Leave this set to 0 for USB sound fobs modified using
; the instructions from usbfob.pdf. Use a setting of
; 1 is for Dingotel/Sph interfaces.
rxboost = yes ; no - 20db attenuator inserted
; yes - 20db attenuator removed
; Set to yes for additional gain if using a low-level receiver output
; CM119B devices are hard coded to rxboost 20dB attenuator removed
rxctcssrelax = 1 ; reduce talkoff from radios w/o CTCSS TX HPF
; Do not change this, leave this as a 1
rxsqhyst = 3000 ; Setting Squelch Hysteresis
; 3000 is the default amount.
; An amount of 500 seems good.
; Then set the squelch to the highest number revealed when doing a "radio
; tune rxsquelch" - running the command multiple (like 20) times in a row.
txctcssdefault = 100.0 ; default TX CTCSS frequency, any frequency permitted
rxctcssfreqs = 100.0,91.5 ; RX CTCSS frequencies list in floating point. must be in table
txctcssfreqs = 100.0,91.5 ; TX CTCSS frequencies list. any frequency permitted. will follow RX CTCSS frequency
rxctcssoverride = no ; Set to yes to start out in carrier squelch mode
carrierfrom = dsp ; no,usb,usbinvert,dsp,vox
; no - no carrier detection at all
; usb - from the COR line on the USB sound fob (Active high)
; usbinvert - from the inverted COR line on the USB sound fob (Active low)
; dsp - from RX noise using DSP techniques
; vox - voice activated from RX audio
; pp - from parallel port (Active high)
; ppinvert - from the inverted parallel port (Active low)
voxhangtime = 2000 ; If carrierfrom=vox, mS to wait detecting RX audio before setting rxcarrierdetect=0
; 2000 is the default but results in 2 seconds of silence added after RX unkey.
; 500 seems more optimal. Lower values may result in unintended drops in RX CD.
ctcssfrom = dsp ; no,usb,usbinvert,dsp
; no - no CTCSS decoding, system will be carrier squelch
; usb - from the CTCSS line on the USB sound fob (Active high)
; usbinvert - from the inverted CTCSS line on the USB sound fob (Active low)
; dsp - CTCSS decoding using RX audio in DSP.
; rxdemod option must be set to flat for this to work.
; pp - from parallel port (Active high)
; ppinvert - from the inverted parallel port (Active low)
rxdemod = flat ; input type from radio: no,speaker,flat
; no - RX audio input not used
; flat - Use RX audio from discriminator (before de-emphasis)
; speaker - use de-emphasized audio
rxsquelchdelay = 30 ; delayline in ms carrier squelch tail eliminator
rxondelay = 0 ; Number of 20mSec intervals following the release of PTT.
; Uncomment and/or adjust for simplex nodes to eliminate "Ping Ponging"
; or "Relay Racing". A positive value here will instruct the usbradio
; driver to ignore the COR line for a specified number of 20mSec
; intervals following the release of PTT. Use this ONLY on simplex
; nodes, and leave commented out for repeaters or other full duplex nodes.
txoffdelay = 0 ; Ignore the receiver for a specified number of 20 millisecond
; intervals after the transmitter unkeys.
; This is useful when setting up a half-duplex link with an existing
; repeater, where you need to ignore the repeater's hangtime.
txboost = no ; Add transmitter audio gain boost: no,yes
txprelim = yes ; Audio processing on left output channel: no,yes
; no - Audio is not pre-emphasized and limited.
; (suitable for use on a microphone input)
; yes - Audio is pre-emphasized and limited.
; (suitable for direct connection to an FM modulator)
txlimonly = yes ; Audio limiting with no pre-emphasis on output channel: no,yes
; no - Audio is not limited.
; yes - Audio is limited.
; Suitable for transmitters with no limiting but with pre-emphasis.
txtoctype = notone ; Transmit tone control type: no,phase,notone
; no - CTCSS tone encoding with no hang time
; phase - encode CTCSS and reverse phase
; AKA ("reverse burst") before unkeying TX
; notone - encode CTCSS and stop sending tone before unkeying TX
; AKA ("chicken burst")
txmixa = composite ; Left channel output: no,voice,tone,composite,auxvoice
; no - Do not output anything
; voice - output voice only
; tone - CTCSS tone only
; composite - voice and tone
; auxvoice - auxiliary voice output at headphone level for monitoring
txmixb = no ; Right channel output: no,voice,tone,composite, auxvoice
; no - Do not output anything
; voice - output voice only
; tone - CTCSS tone only
; composite - voice and tone
; auxvoice - auxiliary voice output at headphone level for monitoring
; Audio filters
rxlpf = 0 ; Receiver Audio Low Pass Filter 0,1 2
; 0 - 3.0 kHz cutoff (Default) value for reduced noise and increased intelligibility.
; 1 - 3.3 kHz cutoff for increased high end, sibilance and brightness.
; 2 - 3.5 kHz cutoff for even more high end, sibilance and brightness.
rxhpf = 0 ; Receiver Audio High Pass Filter 0,1
; 0 - 300 Hz cutoff. (Default) value to reduce sub-audible signals for retransmission and in the receiver speaker.
; 1 - 250 Hz cutoff for additional received and retransmitted bass response.
; recommend using this filter with a CTCSS tone no higher than 186.2 Hz.
txlpf = 0 ; Transmitter Audio Low Pass Filter 0,1
; 0 - 3.0 kHz cutoff. (Default) value.
; 1 - 3.3 kHz cutoff for increased high end, sibilance and brightness.
txhpf = 0 ; Transmitter Audio High Pass Filter 0,1,2
; 0 - 300 Hz cutoff. (Default) value to reduce interference between voice and sub-audible signaling tones and codes.
; 1 - 250 Hz cutoff for increased bass response in transmitted audio.
; 2 - 120 Hz cutoff for special applications requiring additional bass response in transmitted audio.
; Not recommended due to the increased possibility of voice energy interfering with sub-audible signaling
invertptt = no ; Invert PTT: no,yes
; no - ground to transmit
; yes - open to transmit
duplex = 0 ; Duplex 0,1
; 0 - half duplex
; 1 - full duplex
duplex3 = 0 ; duplex 3 gain setting (0 to disable) ???
clipledgpio = 0 ; Enable ADC Clip Detect feature to use a GPIO output (0 to disable).
; Supports URI Clip LED by setting a GPIO (if available) high for 500mS
; when clipping detected. Value = GPIO# to use (GPIO1 recommended)
legacyaudioscaling = no ; If yes, continue to do raw audio sample scaling and clipping, resulting in Tx audio levels increasing
; by 0.83dB and Rx audio levels decreasing by 1.94dB. Should be set to no unless you have an existing
; node with precisely adjusted audio levels and are unable to adjust them. This parameter and associated
; scaling/clipping code will be deleted once existing installs have been able to verify their audio levels
;;; End of node-main template
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;; Configure your nodes here ;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
; Note: there is no need to duplicate entire settings. Only
; place settings that are different than the template.
;
; Note: the device string is automatically found when the
; USB setting "devstr=" is empty.
;
; Note: the interface "tune" settings will be added to the
; per-node settings (below).
[1999](node-main)
;;;;; ASL3 Tune settings ;;;;;
devstr=
rxmixerset=500
txmixaset=500
txmixbset=500
rxvoiceadj=0.5
rxctcssadj=0.5
txctcssadj=200
rxsquelchadj=500

59
asterisk_conf/voter.conf
View file

@ -1,59 +0,0 @@
[general]
;bindaddr = 10.10.10.1 ; specify a particular IP address to bind to (not generally used, defaults to bind to all interfaces)
port = 1667 ; specifies the UDP port to use for incoming connections
buflen = 480 ; specifies the receive buffer length in milliseconds.
; This parameter should be set to the maximum expected network latency,
; plus a little padding (100 milliseconds of padding is a good amount).
; The default is 480 milliseconds, the minimum is 320 milliseconds.
; Buffer length may be specified on a per-stanza and per-client basis, see below.
password = secret_password ; password common to all clients (Main Menu Item 6 - Host Password, on RTCM/VOTER)
utos = y ; Turn on IP TOS for Ubiquiti (ToS is enable by default on the RTCM/VOTER) from the host TO the clients
;sanity = 0 ; disable sanity checking of clients (debug use only) defaults to 1 if not specifically set to 0
;puckit = 1 ; KLUDGE to try and fix random Garmin LVC-18 pucks that may or may not be 1 second off (defaults to 0 if not specified or set to 1)
[1999] ; define the 1999 instance stanza
Main = secret,transmit,master ; master,transmit,adpcm,nulaw,nodeemp,noplfilter,buflen=value,gpsid[=value],prio=value
;
; master - this client is the Master Timing source (the RTCM client that is on the same LAN as the Asterisk server.)
; There can only be 1 Master Timing source per ENTIRE Asterisk server.
;
; secret - password unique to this client
; transmit - this client is intended to have transmit audio sent to it and will have a transmitter connected to it.
; adpcm - this client is to be sent audio in 8000 samples/sec, 4-bit, IMA ADPCM format, rather than default G.711 Mulaw (aka ulaw).
; nulaw - this client is to be sent audio in 4000 samples/second Mulaw (Nulaw, as we call it), rather than the standard 8000 samples/second.
; nodeemp - this client is not to perform de-emphasis of the receiver audio (This is only to be used with non-voting clients). Switches
; the hardware de-emphasis filter OUT on the client
; noplfilter - this client is to not to perform hardware pl filtering of the audio. Switches the hardware PL filter OUT on the client.
; buflen=value - receive buffer length override for this client only.
; gpsid[=value] - This specifies a gps identity to associate with the specified client (as referred in the /etc/asterisk/gps.conf file).
; prio=value - define a specific priority for this client when voting. Lower numbers are higher priority. Default is 0 if nothing specified.
plfilter = y ; use DSP IIR 6 pole High pass filter, 300 Hz corner with 0.5 db ripple (note clients filter PL by default already)
;txctcss = 100.0 ; Transmit CTCSS frequency
;txctcsslevel = 62 ; Transmit CTCSS level (default of 62 if txctcss is supplied, but txctcsslevel is not set)
; Setting CTCSS level adds preemphasis. Assumes modulator input.
;txtoctype = phase ; Transmit tone control type: none,phase,notone
; none - CTCSS tone encoding with no hang time (default)
; phase - encode CTCSS and reverse phase (AKA "reverse burst") before unkeying TX
; notone - encode CTCSS and stop sending tone before unkeying TX (AKA "chicken burst")
;gtxgain = 3.0 ; adjust the audio gain to all transmitters (in db voltage gain)
;hostdeemp = 1 ; force the use of the DSP FIR Integrator providing de-emphasis at 8000 samples/sec. Used with Duplex Mode 3 setting in RTCM/VOTER
thresholds = 255,110=5 ;
; linger=6 ; linger default is 6 if no other value specified
; streams = 67.215.233.178:1667 ; location to send voter data stream for this instance (used only with votemond Java program, which is deprecated)
; isprimary = y ; used in redundant server applications only. must be set on the primary server only
; primary = 10.20.20.1:667,mypaswd ; used in redundant server applications only. must be ONLY set on the secondary server to point to the primary
[1998] ; define another voting instance on this server (for another node). Three simulcast transmitters and four receivers.
; NOTE: there is NO "master" in this instance, since we already have a master timing source on this server defined above
NORTH = password_1,transmit ; transmit/receive site
SOUTH = Password_2,transmit ; transmit/receive site
EAST = password_3,nodeemp ; receive-only site, bypass de-emphasis filter in RTCM
WEST = Password_4,transmit,noplfilter ; transmit/receive site, bypass CTCSS filter in RTCM

1
docker-compose.yml
View file

@ -11,4 +11,5 @@ services:
- ./asl3:/asl3
- ./asterisk_conf:/etc/asterisk
- ./asterisk_cust_sounds:/usr/local/share/asterisk/sounds
- ./asl_backups:/var/asl-backups