; CAPI config
; (1234567 gets replaced by script cfg_asterisk start)
;
; general section
[general]
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 (en/de...)
;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] ; fritzbox 7050/7170 external S0 (or external analog line: experimental)
ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any,
defaultcid=5600XXXX ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
controller=1 ;capi controller number to use (=4: fritzbox 7050/7150 at analog line)
group=1 ;dialout group
softdtmf=off ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=off ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode= ;PBX accountcode to use in CDRs
context=int-so ;context for incoming calls
bridge=no ;native bridging (CAPI line interconnect) if available
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
[ISDN3] ; fritzbox 7050 internal S0
ntmode=no ;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=* ;allow incoming calls to this list of MSNs/DIDs, * = any
defaultcid= ;hier steht die Rufnummer ohne Vorwahl
;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
controller=3 ;capi controller number to use
group=3 ;dialout group
softdtmf=off ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=off ;in addition to softdtmf, you can use relaxed dtmf detection
;accountcode= ;PBX accountcode to use in CDRs
context=capi_in3 ;context for incoming calls
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.
bridge=no ;native bridging (CAPI line interconnect) if available
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
[general]
context=default
bindport=5061
bindaddr=0.0.0.0
localnet=192.168.0.0/255.255.0.0
srvlookup=yes
canreinvite=no ; Typically set to NO if behind NAT
; --------------------------------------------------------------------
;
; hier koennten die Anmeldedaten für VoIP Provider stehen
; dazu kommen wir in einer spaeteren Lektion
;
; --------------------------------------------------------------------
;
; hier kommen die Anmeldekontexte für die SIP Endgeraete7701 - 770x hin
;
[7701]
context=sip7701
callerid="TestSIP 7701" <7701>
host=dynamic
type=friend
user=7701
secret=7701
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
[7702]
context=sip7702
callerid="X-Lite 7702" <7702>
host=dynamic
type=friend
user=7702
secret=7702
username=xlite1
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
; sip external outgoing
[7707]
context=from_fritzbox
callerid="Fritzbox" <7707>
host=dynamic
domain=192.168.123.195
type=friend
user=7707
secret=7707
disallow=all
allow=gsm
allow=ulaw
allow=alaw
[7708]
context=int-so
callerid="Fritzbox" <7708>
host=dynamic
domain=192.168.123.195
type=friend
user=7708
secret=7708
disallow=all
allow=gsm
allow=ulaw
allow=alaw
[globals]
CAPI_CALLERID=5600xxxx
[general]
static=yes
writeprotect=no
; --------------------------------------------------------------------
; Es hat sich als gute Praxis erwiesen, die Inhalte der Datei
; extensions.conf modular aufzubauen. Diese Praxis wollen
; wir auch hier anwenden
;
[lokal]
; Erreichbarkeit der Nebenstellen 7701-7708
; untereinander herstellen 7708 = erste interne a/b-Nebenstelle
exten => _7X,1,NoCDR()
exten => _77XX,1,Dial(SIP/${EXTEN},10,Ttr)
exten => _77XX,2,VoiceMail(${EXTEN},u)
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()
exten => 2900,1,VoiceMailMain(${CALLERID(num)},u)
exten => 2900,2,Hangup()
; --------------------------------------------------------------------
;
; hier kommt der default-Context, in dem alle Geraete in der
; Grundkonfiguration erstmal laufen.
; Alle Geraete koennen sich gegenseitig anrufen
[festnetz_out]
; Raus-Telefonieren
exten => _X.,1,Set(CALLERID(ani)=(${CAPI_CALLERID})
exten => _X.,n,Dial(CAPI/ISDN3/${EXTEN},55,Tt/bd)
[default]
include => lokal
[sip7701]
include => lokal
include => festnetz_out
[sip7702]
include => lokal
include => festnetz_out
[sip7707]
include => lokal
include => festnetz_out
[sip7708]
include => lokal
include => festnetz_out
[capi_in3]
exten => _X.,1,noop(${CALLERID(all)})
exten => _X.,n,Dial(SIP/${EXTEN})
/./var/media/ftp/uStor01/asterisk/bin/start-asterisk.sh > /tmp/debug-out.txt 2>&1
/etc/init.d/rc.S: /tmp/flash/rc.custom: line 1: /var/media/ftp/uStor01/asterisk/bin/start-asterisk.sh: not found
cd /var/media/ftp/uStor01/asterisk/bin
./start-asterisk.sh
/var/mod/sbin/asterisk
Jetzt wo du's sagst, stimmt, da war was.Jetzt werde ich mich noch etwas mit VoiceMail beschäftigen. Habe auch immer noch nicht herausgefunden warum die VoiceMail Zeit nicht mit der Systemzeit übereinstimmt?
all() {
dummy=0
cp -av ../../root-overlay/* ./filesystem
}
cp: Aufruf von stat für „../../root-overlay/*“ nicht möglich: No such file or directory