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|
;
; 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
|
