; ; 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" ' 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