From 31b8fe50815817a3b89c8cb17e5aec3d6286c2f1 Mon Sep 17 00:00:00 2001 From: tilghman Date: Wed, 27 Feb 2008 08:20:15 +0000 Subject: Bring Voicetronix driver up to date with current drivers (closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104502 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/vpb.conf.sample | 336 ++++++++++++++++++++++++++++++++++-------------- 1 file changed, 238 insertions(+), 98 deletions(-) (limited to 'configs') diff --git a/configs/vpb.conf.sample b/configs/vpb.conf.sample index 4a9b0b36a..bcda655de 100644 --- a/configs/vpb.conf.sample +++ b/configs/vpb.conf.sample @@ -1,108 +1,248 @@ ; -; V6PCI/V12PCI config file for VoiceTronix Hardware -; -; Options for [general] section -; -; type = v12pci|v6pci|v4pci -; cards = number of cards -; To use Asterisk indication tones -; indication = 1 -; none,-24db,-18db only for use with OpenLine4 -; ecsuppthres = 0|2048|4096 -; Inter Digit Delay timeout for when collecting DTMF tones for dialling -; from a Station port, in ms -; dtmfidd = 3000 -; To use Asterisk DTMF detection -; ast-dtmf-det=1 -; Used with ast-dtmf-det -; relaxdtmf=1 -; When a native bridge occurs between 2 vpb channels, it will only break -; the connection for '#' and '*' -; break-for-dtmf=no -; Set the maximum period between received rings, default 4000ms -; timer_period_ring=4000 -; -; Options for [interface] section -; board = board_number (1, 2, 3, ...) -; channel = channel_number (1,2,3...) -; mode = fxo|immediate|dialtone -- for type of line and line handling -; context = starting context -; echocancel = on|off (on by default of v4pci, off by default for others) -; callerid = on|off|v23|bell (on => to collect caller ID if available between 1st/2nd rings using vpb functions) -; (v23|bell => collect caller ID using asterisk functions) -; Or for use with FXS channels a '"name" ' format can be used to set the channels CID -; -; UseLoopDrop = 0|1 (enables the use of Loop Drop detection, on by default in -; some cases spurious loop-drops can cause unexpected -; hangup detection) -; -; Gain settings -; txgain => Transmit Software Gain (-12 => 12) -; rxgain => Receive Software Gain (-12 => 12) -; txhwgain => Transmit hardware gain (-12 => 12) -; rxhwgain => Receive Hardware gain (-12 => 12) -; -; These are advanced settings and only mentioned for completeness. -; bal1 => Hybrid balance codec register 1 -; bal2 => Hybrid balance codec register 2 -; bal3 => Hybrid balance codec register 3 -; -; Dial translations - if you want a pause or hook-flash in your dial string -; you can use "w" for pause (wait) or "f" for "hook-flash", eg: -; exten => _9XXX,1,Dial(vpb/g1/ww${EXTEN:${TRUNKMSD}}) +; Voicetronix Voice Processing Board (VPB) telephony interface ; +; Configuration file ; [general] -type = v12pci -;type = v6pci -;type = v4pci -cards = 1 +; +; 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 zapata.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 -board = 0 -echocancel = on - - -; For OpenLine4 cards -;context = demo -;mode = fxo -;channel = 0 -;channel = 1 -;channel = 2 -;channel = 3 - -; For OpenSwith12 with jumpers at factory default -context = demo -mode = fxo -channel = 8 -channel = 9 -channel = 10 -channel = 11 - -context = local -mode = dialtone -channel = 0 -channel = 1 -channel = 2 -channel = 3 -channel = 4 -channel = 5 -channel = 6 -channel = 7 -; -; For OpenSwitch6 -; Note that V6PCI channel numbers start at 7! -;context = demo -;mode = fxo -;channel = 6 -;channel = 7 - -;mode = dialtone -;channel = 8 -;channel = 9 -;channel = 10 -;channel = 11 +; +; 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 -- cgit v1.2.3