diff --git a/asl3-init.sh b/asl3-init.sh index 9be1afc..92d20bf 100644 --- a/asl3-init.sh +++ b/asl3-init.sh @@ -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 diff --git a/asl3/asl3.sh b/asl3/asl3.sh index 8f18379..f2e172c 100755 --- a/asl3/asl3.sh +++ b/asl3/asl3.sh @@ -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 diff --git a/asl_backups/README b/asl_backups/README new file mode 100644 index 0000000..b4da5c4 --- /dev/null +++ b/asl_backups/README @@ -0,0 +1 @@ +This is for backups created by asl-menu diff --git a/asterisk_conf/README b/asterisk_conf/README new file mode 100644 index 0000000..bad4872 --- /dev/null +++ b/asterisk_conf/README @@ -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. diff --git a/asterisk_conf/dnsmgr.conf b/asterisk_conf/dnsmgr.conf deleted file mode 100644 index 31965a8..0000000 --- a/asterisk_conf/dnsmgr.conf +++ /dev/null @@ -1,5 +0,0 @@ -[general] -enable=yes ; enable creation of managed DNS lookups - ; default is 'no' -refreshinterval=300 ; refresh managed DNS lookups every seconds - ; default is 300 (5 minutes) diff --git a/asterisk_conf/echolink.conf b/asterisk_conf/echolink.conf deleted file mode 100644 index c26d915..0000000 --- a/asterisk_conf/echolink.conf +++ /dev/null @@ -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 (request a connect) -; o.dconip (request a disconnect) -; o.rec (turn on/off recording) - -; ipaddr -; port - -#tryinclude custom/echolink.conf diff --git a/asterisk_conf/extensions.conf b/asterisk_conf/extensions.conf deleted file mode 100644 index 725365c..0000000 --- a/asterisk_conf/extensions.conf +++ /dev/null @@ -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 diff --git a/asterisk_conf/iax.conf b/asterisk_conf/iax.conf deleted file mode 100644 index 79483e0..0000000 --- a/asterisk_conf/iax.conf +++ /dev/null @@ -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 = -; secret = -; auth = md5 -; disallow = all -; allow = ulaw -; transfer = no - -#tryinclude custom/iax.conf diff --git a/asterisk_conf/logger.conf b/asterisk_conf/logger.conf deleted file mode 100644 index c651b72..0000000 --- a/asterisk_conf/logger.conf +++ /dev/null @@ -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 diff --git a/asterisk_conf/manager.conf b/asterisk_conf/manager.conf deleted file mode 100644 index b1fb032..0000000 --- a/asterisk_conf/manager.conf +++ /dev/null @@ -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 diff --git a/asterisk_conf/modules.conf b/asterisk_conf/modules.conf deleted file mode 100644 index f57d3cc..0000000 --- a/asterisk_conf/modules.conf +++ /dev/null @@ -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 - diff --git a/asterisk_conf/res_http_media_cache.conf b/asterisk_conf/res_http_media_cache.conf deleted file mode 100644 index 925156b..0000000 --- a/asterisk_conf/res_http_media_cache.conf +++ /dev/null @@ -1,69 +0,0 @@ -; -; 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 diff --git a/asterisk_conf/res_usbradio.conf b/asterisk_conf/res_usbradio.conf deleted file mode 100644 index d4d8030..0000000 --- a/asterisk_conf/res_usbradio.conf +++ /dev/null @@ -1,7 +0,0 @@ -[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) diff --git a/asterisk_conf/rpt.conf b/asterisk_conf/rpt.conf deleted file mode 100644 index d5d47ed..0000000 --- a/asterisk_conf/rpt.conf +++ /dev/null @@ -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,,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. diff --git a/asterisk_conf/rpt_http_registrations.conf b/asterisk_conf/rpt_http_registrations.conf deleted file mode 100644 index 4a6e916..0000000 --- a/asterisk_conf/rpt_http_registrations.conf +++ /dev/null @@ -1,5 +0,0 @@ -[general] -register_interval = 180 ; every 3 minutes - -[registrations] -register => 1234:abcdef@register.allstarlink.org diff --git a/asterisk_conf/simpleusb.conf b/asterisk_conf/simpleusb.conf deleted file mode 100644 index 2d7e40e..0000000 --- a/asterisk_conf/simpleusb.conf +++ /dev/null @@ -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 diff --git a/asterisk_conf/usbradio.conf b/asterisk_conf/usbradio.conf deleted file mode 100644 index ea24982..0000000 --- a/asterisk_conf/usbradio.conf +++ /dev/null @@ -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 diff --git a/asterisk_conf/voter.conf b/asterisk_conf/voter.conf deleted file mode 100644 index 1ffd1f9..0000000 --- a/asterisk_conf/voter.conf +++ /dev/null @@ -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 diff --git a/docker-compose.yml b/docker-compose.yml index 5c8a2de..050b196 100644 --- a/docker-compose.yml +++ b/docker-compose.yml @@ -11,4 +11,5 @@ services: - ./asl3:/asl3 - ./asterisk_conf:/etc/asterisk - ./asterisk_cust_sounds:/usr/local/share/asterisk/sounds + - ./asl_backups:/var/asl-backups