; 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)

; to:

; load => chan_simpleusb.so ;                   CM1xx USB Cards with Radio Interface Channel Driver (No DSP)

 

; SimpleUSB configuration

 

[general]

 

[usb_YOUR-NODE-NUMBER]

 

eeprom = 0              ; EEPROM installed: 0,1

                        ; 0 = no (default)

                        ; 1 = yes

 

hdwtype = 0 ; Leave this set to 0 for USB sound fobs modified using

 

; Receiver parameters

 

rxboost = 0             ; 0 = 20db attenuator inserted, 1= 20db attenuator removed

                        ; Set to 1 for additonal gain if using a low-level receiver output

 

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)

 

ctcssfrom = no ; 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)

 

deemphasis = no         ; enable de-emphasis (input from discriminator)

 

plfilter = yes           ; enable PL filter

 

;rxondelay = 3 ; number of 20ms intervals to hold off receiver turn-on indication

 

 

; 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

 

txboost = 0

 

invertptt = 0           ; Invert PTT: 0,1

                        ; 0 - ground to transmit

                        ; 1 - open to transmit

 

preemphasis = 0 ; 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)

 

duplex = 1              ; Duplex 0,1

                        ; 0 - half duplex

                        ; 1 - full duplex

 

#includeifexists custom/simpleusb.conf

 

/////////////////////////////RPT.CONF////////////////////////////////

 

//////////////////////////////IAX.CONF///////////////////////////////

 

; Inter-Asterisk eXchange driver definition

[general]
; !!! Change this to match your node registration !!!
register=YOUR-NODE-NUMBER:YOUR-ALLSTARLINK-PASSWORD@register.allstarlink.org    ; This must be changed to your node number, password 
; remove the leading ";"

bindport = 4570    ; 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
bindaddr = 0.0.0.0

disallow = all    ; The permitted codecs for outgoing connections 
    ; Audio Quality    Bandwidth
;allow = ulaw    ; best    87 kbps
;allow = adpcm    ; good    55 kbps
;allow = gsm    ; medicore    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

[allstar-public]
type = user
context = allstar-public
auth = md5
secret = allstar
disallow = all
allow = ulaw
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

#includeifexists custom/iax.conf

 

/////////////////////////////MODULES.CONF////////////////////////////

 

;
; Asterisk configuration file
;
; Module Loader configuration file
;
; By default DIAL does NOT load every module, only what is needed

; You can enable or disable any of the asterisk modules
; All modules are compiled and installed.

; To enable a module: load => module_name.so
; To disable a module: noload => module_name.so

; You will want to enable the channel driver modules you will be using.
; There are below in the Channle Driver section
; The most common Channel drivers for app_rpt are:
; chan_echolink.so   echolink channel driver
; chan_simpleusb.so  Simple USB Radio Interface Channel Drive
; chan_usbradio.so   USB Console Channel Driver
; chan_usrp.so       USRP Channel Module
; chan_voter.so      radio Voter channel driver

[modules]

autoload=no

; Applications

noload => app_adsiprog.so ; Asterisk ADSI Programming Application
noload => app_alarmreceiver.so ; Alarm Receiver for Asterisk
noload => app_amd.so ; Answering Machine Detection Application
load => app_authenticate.so ;            Authentication Application
noload => app_cdr.so ; Tell Asterisk to not maintain a CDR for
noload => app_chanisavail.so ; Check channel availability
noload => app_channelredirect.so ; Channel Redirect
noload => app_chanspy.so ; Listen to the audio of an active channel
noload => app_controlplayback.so ; Control Playback Application
noload => app_dahdibarge.so ; Barge in on channel application
noload => app_dahdiras.so ; DAHDI RAS Application
noload => app_dahdiscan.so ; Scan Zap channels application
noload => app_db.so ; Database Access Functions
load => app_dial.so ; Dialing Application
noload => app_dictate.so ; Virtual Dictation Machine
noload => app_directed_pickup.so ; Directed Call Pickup Application
noload => app_directory.so ; Extension Directory
noload => app_disa.so ; DISA (Direct Inward System Access) Appli
noload => app_dumpchan.so ; Dump Info About The Calling Channel
noload => app_echo.so ; Simple Echo Application
load => app_exec.so ; Executes dialplan applications
noload => app_externalivr.so ; External IVR Interface Application
noload => app_festival.so ; Simple Festival Interface
noload => app_flash.so ; Flash channel application
noload => app_followme.so ; Find-Me/Follow-Me Application
noload => app_forkcdr.so ; Fork The CDR into 2 separate entities
noload => app_getcpeid.so ; Get ADSI CPE ID
noload => app_gps.so ; GPS interface module
noload => app_hasnewvoicemail.so ; Indicator for whether a voice mailbox ha
noload => app_ices.so ; Encode and Stream via icecast and ices
noload => app_image.so ; Image Transmission Application
noload => app_lookupblacklist.so ; Look up Caller*ID name/number from black
noload => app_lookupcidname.so ; Look up CallerID Name from local databas
load => app_macro.so ; Extension Macros
noload => app_meetme.so ; MeetMe conference bridge
noload => app_milliwatt.so ; Digital Milliwatt (mu-law) Test Applicat
noload => app_mixmonitor.so ; Mixed Audio Monitoring Application
noload => app_morsecode.so ; Morse code
noload => app_mp3.so ; Silly MP3 Application
noload => app_nbscat.so ; Silly NBS Stream Application
noload => app_page.so ; Page Multiple Phones
noload => app_parkandannounce.so ; Call Parking and Announce Application
load => app_playback.so ; Sound File Playback Application
noload => app_privacy.so ; Require phone number to be entered, if n
noload => app_queue.so ; True Call Queueing
noload => app_radbridge.so ; Radio Bridging interface module
noload => app_random.so ; Random goto
noload => app_readfile.so ; Stores output of file into a variable
noload => app_read.so ; Read Variable Application
noload => app_realtime.so ; Realtime Data Lookup/Rewrite
noload => app_record.so ; Trivial Record Application
load => app_rpt.so ; Radio Repeater/Remote Base Application
noload => app_sayunixtime.so ; Say time
noload => app_senddtmf.so ; Send DTMF digits Application
load => app_sendtext.so ; Send Text Applications
noload => app_setcallerid.so ; Set CallerID Application
noload => app_setcdruserfield.so ; CDR user field apps
noload => app_settransfercapability.so ; Set ISDN Transfer Capability
noload => app_sms.so ; SMS/PSTN handler
noload => app_softhangup.so ; Hangs up the requested channel
noload => app_speech_utils.so ; Dialplan Speech Applications
noload => app_stack.so ; Stack Routines
load => app_system.so ; Generic System() application
load => app_talkdetect.so ; Playback with Talk Detection
noload => app_test.so ; Interface Test Application
load => app_transfer.so ; Transfer
noload => app_url.so ; Send URL Applications
noload => app_userevent.so ; Custom User Event Application
noload => app_verbose.so ; Send verbose output
noload => app_voicemail.so ; Comedian Mail (Voicemail System)
noload => app_waitforring.so ; Waits until first ring after time
noload => app_waitforsilence.so ; Wait For Silence
noload => app_while.so ; While Loops and Conditional Execution
noload => app_zapateller.so ; Block Telemarketers with Special Informa

; CDR

noload => cdr_csv.so ; Comma Separated Values CDR Backend
noload => cdr_custom.so ; Customizable Comma Separated Values CDR
noload => cdr_manager.so ; Asterisk Manager Interface CDR Backend

; Channels

noload => chan_agent.so ; Agent Proxy Channel
noload => chan_alsa.so ; ALSA Console Channel Driver
noload => chan_beagle.so ; Beagleboard Radio Interface Channel Driver
load => chan_dahdi.so ; DAHDI Telephony
noload => chan_echolink.so ; echolink Channel Driver
noload => chan_features.so ; Feature Proxy Channel
noload => chan_gtalk.so ; Gtalk Channel Driver
load => chan_iax2.so ; Inter Asterisk eXchange (Ver 2)
load => chan_local.so ; Local Proxy Channel (Note: used internal
noload => chan_oss.so ; Channel driver for OSS sound cards
noload => chan_phone.so ; Generic Linux Telephony Interface driver
noload => chan_pi.so ; DMK Engineering "PITA" Board on Rpi2/3 Channel Driver
load => chan_simpleusb.so ; CM1xx USB Cards with Radio Interface Channel Driver (No DSP)
noload => chan_sip.so ; Session Initiation Protocol (SIP)
noload => chan_tlb.so ; TheLinkBox Channel Driver
noload => chan_usbradio.so ; CM1xx USB Cards with Radio Interface Channel Driver (DSP)
noload => chan_usrp.so ; GNU Radio interface USRP Channel Driver
noload => chan_voter.so ; Radio Voter 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 Translato
load => codec_g726.so ; ITU G.726-32kbps G726 Transcoder
load => codec_gsm.so ; GSM Coder/Decoder
load => codec_ulaw.so ; mu-Law Coder/Decoder
noload => codec_ilbc.so ; http://en.wikipedia.org/wiki/Internet_Low_Bitrate_Codec

; Formats

load => format_g723.so ; G.723.1 Simple Timestamp File Format
load => format_g726.so ; Raw G.726 (16/24/32/40kbps) data
load => format_g729.so ; Raw G729 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
noload => format_jpeg.so ; JPEG (Joint Picture Experts Group) Image
load => format_pcm.so ; Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.
load => format_sln.so ; Raw Signed Linear Audio support (SLN)
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 format (8000Hz Signed Line

; Functions

load => func_base64.so ; base64 encode/decode dialplan functions
load => func_callerid.so ; Caller ID related dialplan function
load => func_cdr.so ; CDR dialplan function
load => func_channel.so ; Channel information dialplan function
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 functi
load => func_enum.so ; ENUM related dialplan functions
load => func_env.so ; Environment/filesystem dialplan function
load => func_global.so ; Global variable dialplan functions
load => func_groupcount.so ; Channel group dialplan functions
load => func_language.so ; Channel language dialplan function
load => func_logic.so ; Logical dialplan functions
load => func_math.so ; Mathematical dialplan function
load => func_md5.so ; MD5 digest dialplan functions
load => func_moh.so ; Music-on-hold dialplan function
load => func_rand.so ; Random number dialplan function
load => func_realtime.so ; Read/Write values from a RealTime reposi
noload => func_sha1.so ; SHA-1 computation dialplan function
noload => func_strings.so ; String handling dialplan functions
noload => func_timeout.so ; Channel timeout dialplan functions
noload => func_uri.so ; URI encode/decode dialplan functions

; PBX

noload => pbx_ael.so ; Asterisk Extension Language Compiler
load => pbx_config.so ; Text Extension Configuration
noload => pbx_dundi.so ; Distributed Universal Number Discovery (
noload => pbx_loopback.so ; Loopback Switch
noload => pbx_realtime.so ; Realtime Switch
noload => pbx_spool.so ; Outgoing Spool Support

; Resources

load => res_adsi.so ; ADSI Resource
noload => res_agi.so ; Asterisk Gateway Interface (AGI)
noload => res_clioriginate.so ; Call origination from the CLI
noload => res_convert.so ; File format conversion CLI command
load => res_crypto.so ; Cryptographic Digital Signatures
load => res_features.so ; Call Features Resource
load => res_indications.so ; Indications Resource
noload => res_jabber.so ; AJI - Asterisk Jabber Interface
noload => res_monitor.so ; Call Monitoring Resource
noload => res_musiconhold.so ; Music On Hold Resource
load => res_smdi.so ; Simplified Message Desk Interface (SMDI)
noload => res_snmp.so ; SNMP [Sub]Agent for Asterisk
noload => res_speech.so ; Generic Speech Recognition API

[global]

 

/////////////////////////////MANAGER.CONF////////////////////////////

 

;
; Asterisk Call Management support
;

; By default asterisk will listen on localhost only. 
[general]
enabled = yes
port = 4570
;bindaddr = 192.168.42.3
binadddr = 0.0.0.0

[admin]
secret = (YOUR-SECRET-PASSWORD)
read = all,system,call,log,verbose,command,agent,user,config
write = all,system,call,log,verbose,command,agent,user,config