Hallo...
seit mitte Dezember letzten Jahres plage ich mich inzwischen mit meiner Uralten Fritz!Card PCI herum und hab es inzwischen geschafft vom SIP Phone über asterisk nach ausen telefonieren zu können. Intern übermittelt Asterisk sogar die korrekte MSN 19, nach aussen hängt ne Eumex XI 520 dazwischen. Also folgender aufbau:
Außenwelt <-> Eumex XI 520 <-> Fritz!Card PCI UND einige ISDN Telefone am S0-Bus UND einige Analoge Telefone <-> Asterisk 1.6.2.0 auf Debian Lenny kernel 2.6.26-2-amd64 mit Capi 2.0/chan_capi HEAD <-> einige Softphones mit SIP-Accounts am Asterisk
Ich kann jetzt vom Softphone aus eines der analogen Telefone oder der ISDN Telefone an der Anlage genau so gut wie alle anderen Telefone der aussenwelt anrufen.
Rufe ich nun den Asterisk server (mit der in der capi.conf konfigurierten MSN 19) an, so zeigt dieser eine Reaktion in der Konsole ab verbosity 2 an, verarbeitet den anruf aber nicht wie gewollt weiter.
Hier meine Konfigurationsdateien:
capi.conf:
extensions.conf:
und die Ausgabe in der Asterisk konsole bei verbosity 10 wenn man die msn 19 anwählt:
Es wäre super wenn mir jemand helfen könnte, wenn ihr noch Informationen braucht fragt einfach
Grüße: Merhoc
seit mitte Dezember letzten Jahres plage ich mich inzwischen mit meiner Uralten Fritz!Card PCI herum und hab es inzwischen geschafft vom SIP Phone über asterisk nach ausen telefonieren zu können. Intern übermittelt Asterisk sogar die korrekte MSN 19, nach aussen hängt ne Eumex XI 520 dazwischen. Also folgender aufbau:
Außenwelt <-> Eumex XI 520 <-> Fritz!Card PCI UND einige ISDN Telefone am S0-Bus UND einige Analoge Telefone <-> Asterisk 1.6.2.0 auf Debian Lenny kernel 2.6.26-2-amd64 mit Capi 2.0/chan_capi HEAD <-> einige Softphones mit SIP-Accounts am Asterisk
Ich kann jetzt vom Softphone aus eines der analogen Telefone oder der ISDN Telefone an der Anlage genau so gut wie alle anderen Telefone der aussenwelt anrufen.
Rufe ich nun den Asterisk server (mit der in der capi.conf konfigurierten MSN 19) an, so zeigt dieser eine Reaktion in der Konsole ab verbosity 2 an, verarbeitet den anruf aber nicht wie gewollt weiter.
Hier meine Konfigurationsdateien:
capi.conf:
Code:
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0 ; or for example "+49"
internationalprefix=00 ; or for example "+"
;subscriberprefix=+4969 ; prefix including area code (some lines need this)
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 the 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=19 ;allow incoming calls to this list of MSNs/DIDs, * = any
defaultcid=19 ;set a default caller ID to that interface for dial-out,
;this caller ID will be used when the 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 the 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')
faxdetecttime=0 ;Only detect faxes during the first 'n' seconds of the call.
;(default '0' meaning for the whole duration of the call)
accountcode= ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or 'documentation')
context=isdn-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 do not 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. Disable it before you start recording voicemail
;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
echocancel=yes ;Dialogic(R) Diva(R) (CAPI) echo cancellation (yes=g165)
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
;echocancelpath=1;Dialogic(R) Diva(R) (CAPI) echo cancellation path
;(possible values: default '1' - E.1/T.1/S0, '2' - IP, '3' - both)
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)
;transfergroup=1 ;Controller(s) where a transfer on native bridge is allowed to.
language=de ;set language for this device (overwrites default language)
;disallow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)
;allow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards 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
extensions.conf:
Code:
[default]
;switch => Realtime/@sipuserdialplan
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Playback(yeah)
exten => 1001,4,Hangup()
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,VoiceMail(2000,u)
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,VoiceMail(2001,u)
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,VoiceMail(2002,u)
exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)
exten => _**X.,1,Dial(CAPI/g1/19:${EXTEN})
exten => _0X.,1,Dial(CAPI/g1/19:${EXTEN})
[isdn-in]
exten => 19,1,Dial(SIP/2000)
und die Ausgabe in der Asterisk konsole bei verbosity 10 wenn man die msn 19 anwählt:
Code:
CAPI: ApplId=0x0002 Command=0x02 SubCommand=0x82 MsgNum=0x0026 NCCI=0x00000301
CONNECT_IND ID=002 #0x0026 LEN=0037
Controller/PLCI/NCCI = 0x301
CIPValue = 0x10
CalledPartyNumber = <80>19
CallingPartyNumber = <01 80 2a 2a>18
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = <80 90 a3>
LLC = default
HLC = <91 81>
AdditionalInfo = default
-- CONNECT_IND (PLCI=0x301,DID=19,CID=**18,CIP=0x10,CONTROLLER=0x1)
> ISDN1#02: msn='19' DNID='19' MSN
== ISDN1#02: setting format alaw - 0x8 (alaw)
== ISDN1#02: Incoming call '**18' -> '19'
CAPI: ApplId=0x0002 Command=0x08 SubCommand=0x82 MsgNum=0x0027 NCCI=0x00000301
INFO_IND ID=002 #0x0027 LEN=0018
Controller/PLCI/NCCI = 0x301
InfoNumber = 0x70
InfoElement = <80>19
INFO_RESP ID=002 #0x0027 LEN=0012
Controller/PLCI/NCCI = 0x301
-- ISDN1#02: info element CALLED PARTY NUMBER
> ISDN1#02: INFO_IND DID digits not used in this state.
CAPI: ApplId=0x0002 Command=0x08 SubCommand=0x82 MsgNum=0x0028 NCCI=0x00000301
INFO_IND ID=002 #0x0028 LEN=0016
Controller/PLCI/NCCI = 0x301
InfoNumber = 0x18
InfoElement = <89>
INFO_RESP ID=002 #0x0028 LEN=0012
Controller/PLCI/NCCI = 0x301
-- ISDN1#02: info element CHANNEL IDENTIFICATION 89
CAPI: ApplId=0x0002 Command=0x08 SubCommand=0x82 MsgNum=0x0029 NCCI=0x00000301
INFO_IND ID=002 #0x0029 LEN=0018
Controller/PLCI/NCCI = 0x301
InfoNumber = 0x70
InfoElement = <80>19
INFO_RESP ID=002 #0x0029 LEN=0012
Controller/PLCI/NCCI = 0x301
-- ISDN1#02: info element CALLED PARTY NUMBER
> ISDN1#02: INFO_IND DID digits not used in this state.
CAPI: ApplId=0x0002 Command=0x08 SubCommand=0x82 MsgNum=0x002a NCCI=0x00000301
INFO_IND ID=002 #0x002a LEN=0016
Controller/PLCI/NCCI = 0x301
InfoNumber = 0x18
InfoElement = <89>
INFO_RESP ID=002 #0x002a LEN=0012
Controller/PLCI/NCCI = 0x301
-- ISDN1#02: info element CHANNEL IDENTIFICATION 89
CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x82 MsgNum=0x002b NCCI=0x00000301
DISCONNECT_IND ID=002 #0x002b LEN=0014
Controller/PLCI/NCCI = 0x301
Reason = 0x0
DISCONNECT_RESP ID=002 #0x002b LEN=0012
Controller/PLCI/NCCI = 0x301
-- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup.
> CAPI/ISDN1#02/19-5: set channel task to 1
== ISDN1#02: CAPI Hangingup for PLCI=0x301 in state 4
== ISDN1#02: Interface cleanup PLCI=0x301
> chan_capi devicestate requested for ISDN1#02/19 is 'Not in use'
Es wäre super wenn mir jemand helfen könnte, wenn ihr noch Informationen braucht fragt einfach
Grüße: Merhoc