;
; CAPI config
;
;
; general section
[general]
chan_capi.so=yes
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=0231******* ;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number of this interface/port
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or
;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')
accountcode= ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or 'documentation')
context=capi-in ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and the PBX may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation (yes=g165)
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting (default=0 for maximum)
;echocancelnlp=1 ;activate non-linear-processing; this improves echo cancel ratio, but might
;incorporate variable gain in the signal path.
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;PBX call group
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)
;language=de ;set language for this device (overwrites default language)
;disallow=all ;RTP codec selection (valid with Eicon DIVA Server only)
;allow=all ;RTP codec selection (valid with Eicon DIVA Server only)
devices=2 ;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
;qsig=1 ;enable use of Q.SIG extensions. ECMA Variant
;qsig_prnum=1234 ;enable inbound bridging - this should be an QSIG-network-wide unique number[general]
static=yes
chan_capi.so=yes
writeprotect=no
; --------------------------------------------------------------------
;[echotest]
;exten => 81,1,answer
;exten => 81,2,wait,1
;exten => 81,3,playback,demo-echotest
;exten => 81,4,echo
;exten => 81,5,playback,demo-echodone
;exten => 81,6,hangup
[mailbox]
exten => 80,1,VoiceMailMain(${CALLERID(num)},s)
[local]
exten => _3X,1,NoCDR()
exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr
[arcor_out]
exten => _0.,1,Dial,SIP/${EXTEN}@arcor|45|Ttr
[ankommend]
exten => 0231*****,1,Dial(SIP/30|30|r)
exten => 0231*****,n,Playback(/var/lib/asterisk/mysounds/ansage)
exten => 0231*****,n,voicemail,s30
exten => 0231*****,n,Hangup
[eingehend]
exten => 0231****,1,Dial(SIP/30,20)
exten => 0231****,n,VoiceMail(30)
;------------------------------------------------------------------
[default]
;include => echotest
include => mailbox
include => local
include => arcor_out
include => ankommend
include => eingehendcontext=capi-in ;context for incoming calls
