249 lines
5.8 KiB
Plaintext
249 lines
5.8 KiB
Plaintext
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; Voicetronix Voice Processing Board (VPB) telephony interface
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;
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; Configuration file
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;
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[general]
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;
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; Total number of Voicetronix cards in this machine
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;
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cards=0
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;
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; Which indication functions to use
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; 1 = use Asterisk functions
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; 0 = use VPB functions
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;
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indication=1
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;
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; Echo Canceller suppression threshold
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; 0 = no suppression threshold
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; 2048 = -18dB
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; 4096 = -24dB
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;
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;ecsuppthres=0
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;
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; Inter-digit delay timeout, used when collecting DTMF tones for dialling
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; from a station port. Measured in milliseconds.
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;
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dtmfidd=3000
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;
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; How to play DTMF tones
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; any value = use Asterisk functions
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; commented out = use VPB functions
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;
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;ast-dtmf=1
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;
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; How to detect DTMF tones
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; any value = use Asterisk functions
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; commented out = use VPB functions
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;
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; NOTE: this setting is currently broken, and uncommenting it will
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; stop dialling from working. Any volunteers to fix it?
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;ast-dtmf-det=1
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;
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; Use relaxed DTMF detection (ignored unless ast-dtmf-det is set)
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;
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relaxdtmf=1
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;
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; When we do a native bridge between two VPB channels:
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; yes = only break the connection for '#' and '*'
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; no = break the connection for any DTMF
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;
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; NOTE: this is currently broken, and setting to no will segfault
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; Asterisk while dialling. Any volunteers to fix it?
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;
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break-for-dtmf=yes
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;
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; The maximum period between received rings. Measures in milliseconds.
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;
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timer_period_ring=4000
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[interfaces]
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;
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; Default language
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;
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language=en
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;
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; Default context
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;
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context=public
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;
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; Echo cancellation
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; off = no not use echo cancellation
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; on = use echo cancellation
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;
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echocancel=off
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;
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; Caller ID routines/signalling
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; For FXO ports, select one of:
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; on = collect caller ID between 1st/2nd rings using VPB routines
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; off = do not use caller ID
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; bell = bell202 as used in US, using Asterisk's caller ID routines
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; v23 = v23 as used in the UK, using Asterisk's caller ID routines
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; For FXS ports, set the channel's CID in '"name" <number>' format
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;
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; NOTE that other caller ID standards are supported in Asterisk, but are
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; not yet active in chan_vpb. It should be reasonably trivial to add
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; support for the other standards (see the default chan_dahdi.conf for a
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; list of them) that Asterisk already handles.
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;
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callerid=bell
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;
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; Use a polarity reversal as the trigger for the start of caller ID,
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; rather than triggering after the first ring.
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;
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usepolaritycid=0
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;
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; Use loop drop to detect the end of a call. On by default, but if you
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; experience unexpected hangups, try turning it off.
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;
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useloopdrop=1
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;
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; Use in-kernel bridging. This will generally give lower delay audio if
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; bridging between two VPB channels. It will not affect bridging
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; between VPB channels and other technologies.
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;
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usenativebridge=1
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;
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; Software transmit and receive gain. Adjusting these will change the
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; volume of audio files that are played (tx) and recorded (rx). It will
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; _not_ affect audio between channels in a native bridge. It will,
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; however, affect the volume of audio between VPB channels and channels
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; using other technologies (such as VoIP channels). Usually it's best to
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; leave these as they are. If you're looking to get rid of echo, the
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; first thing to do is match your line impedance with the bal1/bal2/bal3
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; settings.
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;
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;txgain=0.0
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;rxgain=0.0
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;
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; Hardware transmit and receive gain. Adjusting these will change the
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; volume of all audio on a channel. The allowed range of settings is
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; -12.0 to 12.0 (measured in dB).
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;
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;txhwgain=0.0
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;rxhwgain=0.0
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;
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; Balance register settings, for matching the impedance of the card to
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; that of the connected equipment. Only relevant for OpenLine and
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; OpenSwitch series cards. Values should be in the range 0 - 255.
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;
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; We (Voicetronix) have determined the best codec balance values for
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; standard interfaces based on their US, Australian and European
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; specifications, shown below.
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;
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; US (600 ohm)
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;bal1=0xf8
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;bal2=0x1a
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;bal3=0x0c
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;
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; Australia (complex impedance)
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;bal1=0xf0
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;bal2=0x5d
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;bal3=0x79
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;
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; Europe (CTR-21)
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;bal1=0xf0
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;bal2=0x6e
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;bal3=0x75
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;
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; Logical groups can be assigned to allow outgoing rollover. Groups range
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; from 0 to 63, and multiple groups can be specified.
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;
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group=1
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;
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; Ring groups (a.k.a. call groups) and pickup groups. If a phone is
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; ringing and it is a member of a group which is one of your pickup
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; groups, then you can answer it by picking up and dialling *8#. For
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; simple offices, just make these both the same. Groups range from 0 to
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; 63.
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;
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callgroup=1
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pickupgroup=1
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;
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; If we haven't had a "grunt" (voice activity detection) for this many
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; seconds, then we hang up the line due to inactivity. Default is one
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; hour.
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;
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grunttimeout=3600
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;
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; Type of line and line handling. This setting will usually be overridden
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; on a per channel basis. Valid settings are:
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; fxo = this is an FXO port
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; immediate = this is an FXS port, with no dialtone or dialling
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; required (ie it is a "hotline")
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; dialtone = this is an FXS port, providing dialtone and dialling
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;
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mode=immediate
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;-------------------------------------------------------------------------
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; Channel definitions
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;
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; Each channel inherits the settings specified above, unless the are
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; overridden. As a minimum, the board number and channel number must be
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; set, starting from 0 for the first board, and for the channels on each
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; board. For example, board 0, channels 0 to 11, then board 1, channels
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; 0 to 11 for two OpenSwitch12 cards.
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;
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;
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; First board is an OpenSwitch12 card (jumpers at factory defaults)
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;
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;board=0
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;
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;mode=dialtone
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;context=from-handset
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;group=1
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;channel=0
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;channel=1
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;channel=2
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;channel=3
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;channel=4
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;channel=5
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;channel=6
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;channel=7
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;
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;mode=fxo
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;context=from-pstn
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;group=2
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;channel=8
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;channel=9
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;channel=10
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;channel=11
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;
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; Second board is an OpenLine4
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;
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;board=1
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;
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;mode=fxo
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;group=2
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;context=from-pstn
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;channel=0
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;channel=1
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;channel=2
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;channel=3
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