jw-asterisk-docker/asterisk/vpb.conf

;
; Voicetronix Voice Processing Board (VPB) telephony interface
;
; Configuration file
;

[general]
;
; Total number of Voicetronix cards in this machine
;
cards=0

;
; Which indication functions to use
;    1 = use Asterisk functions
;    0 = use VPB functions
;
indication=1

;
; Echo Canceller suppression threshold
;    0    = no suppression threshold
;    2048 = -18dB
;    4096 = -24dB
;
;ecsuppthres=0

;
; Inter-digit delay timeout, used when collecting DTMF tones for dialling
; from a station port.  Measured in milliseconds.
;
dtmfidd=3000

;
; How to play DTMF tones
;    any value     = use Asterisk functions
;    commented out = use VPB functions
;
;ast-dtmf=1

;
; How to detect DTMF tones
;    any value     = use Asterisk functions
;    commented out = use VPB functions
;
; NOTE: this setting is currently broken, and uncommenting it will
; stop dialling from working.  Any volunteers to fix it?
;ast-dtmf-det=1

;
; Use relaxed DTMF detection (ignored unless ast-dtmf-det is set)
;
relaxdtmf=1

;
; When we do a native bridge between two VPB channels:
;    yes = only break the connection for '#' and '*'
;    no  = break the connection for any DTMF
;
; NOTE: this is currently broken, and setting to no will segfault
; Asterisk while dialling.  Any volunteers to fix it?
;
break-for-dtmf=yes

;
; The maximum period between received rings.  Measures in milliseconds.
;
timer_period_ring=4000


[interfaces]
;
; Default language
;
language=en

;
; Default context
;
context=default

;
; Echo cancellation
;     off  = no not use echo cancellation
;     on   = use echo cancellation
;
echocancel=off

;
; Caller ID routines/signalling
;   For FXO ports, select one of:
;     on   = collect caller ID between 1st/2nd rings using VPB routines
;     off  = do not use caller ID
;     bell = bell202 as used in US, using Asterisk's caller ID routines
;     v23  = v23 as used in the UK, using Asterisk's caller ID routines
;   For FXS ports, set the channel's CID in '"name" <number>' format
;
; NOTE that other caller ID standards are supported in Asterisk, but are
; not yet active in chan_vpb.  It should be reasonably trivial to add
; support for the other standards (see the default chan_dahdi.conf for a
; list of them) that Asterisk already handles.
;
callerid=bell

;
; Use a polarity reversal as the trigger for the start of caller ID,
; rather than triggering after the first ring.
;
usepolaritycid=0

;
; Use loop drop to detect the end of a call.  On by default, but if you
; experience unexpected hangups, try turning it off.
;
useloopdrop=1

;
; Use in-kernel bridging.  This will generally give lower delay audio if
; bridging between two VPB channels.  It will not affect bridging
; between VPB channels and other technologies.
;
usenativebridge=1

;
; Software transmit and receive gain.  Adjusting these will change the
; volume of audio files that are played (tx) and recorded (rx).  It will
; _not_ affect audio between channels in a native bridge.  It will,
; however, affect the volume of audio between VPB channels and channels
; using other technologies (such as VoIP channels).  Usually it's best to
; leave these as they are.  If you're looking to get rid of echo, the
; first thing to do is match your line impedance with the bal1/bal2/bal3
; settings.
;
;txgain=0.0
;rxgain=0.0

;
; Hardware transmit and receive gain.  Adjusting these will change the
; volume of all audio on a channel.  The allowed range of settings is
; -12.0 to 12.0 (measured in dB).
;
;txhwgain=0.0
;rxhwgain=0.0

;
; Balance register settings, for matching the impedance of the card to
; that of the connected equipment.  Only relevant for OpenLine and
; OpenSwitch series cards.  Values should be in the range 0 - 255.
;
; We (Voicetronix) have determined the best codec balance values for
; standard interfaces based on their US, Australian and European
; specifications, shown below.
;
; US (600 ohm)
;bal1=0xf8
;bal2=0x1a
;bal3=0x0c
;
; Australia (complex impedance)
;bal1=0xf0
;bal2=0x5d
;bal3=0x79
;
; Europe (CTR-21)
;bal1=0xf0
;bal2=0x6e
;bal3=0x75

;
; Logical groups can be assigned to allow outgoing rollover.  Groups range
; from 0 to 63, and multiple groups can be specified.
;
group=1

;
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is
; ringing and it is a member of a group which is one of your pickup
; groups, then you can answer it by picking up and dialling *8#.  For
; simple offices, just make these both the same.  Groups range from 0 to
; 63.
;
callgroup=1
pickupgroup=1

;
; If we haven't had a "grunt" (voice activity detection) for this many
; seconds, then we hang up the line due to inactivity.  Default is one
; hour.
;
grunttimeout=3600

;
; Type of line and line handling.  This setting will usually be overridden
; on a per channel basis.  Valid settings are:
;     fxo       = this is an FXO port
;     immediate = this is an FXS port, with no dialtone or dialling
;                   required (ie it is a "hotline")
;     dialtone  = this is an FXS port, providing dialtone and dialling
;
mode=immediate

;-------------------------------------------------------------------------
; Channel definitions
;
; Each channel inherits the settings specified above, unless the are
; overridden.  As a minimum, the board number and channel number must be
; set, starting from 0 for the first board, and for the channels on each
; board.  For example, board 0, channels 0 to 11, then board 1, channels
; 0 to 11 for two OpenSwitch12 cards.
;

;
; First board is an OpenSwitch12 card (jumpers at factory defaults)
;
;board=0
;
;mode=dialtone
;context=from-handset
;group=1
;channel=0
;channel=1
;channel=2
;channel=3
;channel=4
;channel=5
;channel=6
;channel=7
;
;mode=fxo
;context=from-pstn
;group=2
;channel=8
;channel=9
;channel=10
;channel=11

;
; Second board is an OpenLine4
;
;board=1
;
;mode=fxo
;group=2
;context=from-pstn
;channel=0
;channel=1
;channel=2
;channel=3