AVM C4 Anlagenanschluss und PBX Problem

segelfreak

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Hallo an Alle!

Ich habe hier aktuell ein Problem bei der ricvhtigen Konfiguration einer AVM C4 Karte am Anlagenanschluss mit nachgelagerter Hicom TK-Anlage.

Der Betrieb der C4 am Anlagenanschluß mit zwei "Leitungen" (also 4 Kanälen)funktioniert komplett mit der Asterisk.
Soweit also alles prima.

Nun möchte ich gerne eine Hicom an die verbleibenen 2 Ports hängen.

Ein Cross-Over Kabel habe ich mit gebastelt und auch an der Hicom am internen S0 Port erfolgreich getestet. Über einen eingeschleiften NTBA habe ich sogar ein ISDN Telefon testen können.

Wenn ich nun dieses an die C4 anschließe, bekomme ich nur eine Störung. Ein Test mit einer FritzCard zeigt einen Protokollfehler Ebene 1. (Das Ganze wieder an der Hicom - interner S0 - ist wieder ok.

Irgendetwas läuft also offenbar noch schief - vermutlich mit meiner config?

Die sieht wie folgt aus:

/etc/isdn/capi.conf
Code:
# card		file			proto	io	irq	mem	cardnr	options
#b1isa		b1.t4			DSS1	0x150	7	-	-	P2P
#b1pci		b1.t4	[12~		DSS1	-	-	-	-

c4		/etc/isdn/c4.bin	DSS1	-	-	-	-	P2P
c4		-		DSS1	-	-	-	-	P2P
c4		-		DSS1	-	-	-	-	
c4		-		DSS1	-	-	-	-	

#c2		c2.bin		DSS1	-	-	-	-
#c2		-		DSS1	-	-	-	-
#t1isa		t1.t4		DSS1	0x340	9	-	0
#t1pci		t1.t4		DSS1	-	-	-	-
#fcpci		-		-	-	-	-	-
#fcpcmcia	-		-	-	-	-	-
#fcusb		-		-	-	-	-	-
#fxusb		-		-	-	-	-	-
#fcclassic	-		-	0x150	10	-	-
#fcdsl		fdslbase.bin	DSS1	-	-	-	-
#fcdsl2		fds2base.bin	DSS1	-	-	-	-
#fcdslsl	fdssbase.bin	DSS1	-	-	-	-
#fcdslslusb	fdlubase.frm	DSS1	-	-	-	-
#fcdslusba	fdlabase.frm	DSS1	-	-	-	-
#fcdslusb2	fds2base.frm	DSS1	-	-	-	-
#fcdslusb	fdsubase.frm	DSS1	-	-	-	-

Kontrolle capiinfo:
Code:
Number of Controllers : 4
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-06  (49.22)
Serial Number: 0900068
BChannels: 2
Global Options: 0x00000039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x4000011f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x00000b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x800000bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  39000000
  1f010040
  1b0b0000
  bf000080
  00000000 00000000 00000000 00000000 00000000 00000000
  01000001 00010000 00000000 00000000 00000000

Supplementary services support: 0x000003ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

Controller 2:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-06  (49.22)
Serial Number: 0900068
BChannels: 2
Global Options: 0x00000039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x4000011f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x00000b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x800000bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  39000000
  1f010040
  1b0b0000
  bf000080
  00000000 00000000 00000000 00000000 00000000 00000000
  01000001 00010000 00000000 00000000 00000000

Supplementary services support: 0x000003ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

Controller 3:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-06  (49.22)
Serial Number: 0900068
BChannels: 2
Global Options: 0x00000039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x4000011f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x00000b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x800000bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  39000000
  1f010040
  1b0b0000
  bf000080
  00000000 00000000 00000000 00000000 00000000 00000000
  01000001 00020000 00000000 00000000 00000000

Supplementary services support: 0x000003ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

Controller 4:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-06  (49.22)
Serial Number: 0900068
BChannels: 2
Global Options: 0x00000039
   internal controller supported
   DTMF supported
   Supplementary Services supported
   channel allocation supported (leased lines)
B1 protocols support: 0x4000011f
   64 kbit/s with HDLC framing
   64 kbit/s bit-transparent operation
   V.110 asynconous operation with start/stop byte framing
   V.110 synconous operation with HDLC framing
   T.30 modem for fax group 3
   Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x00000b1b
   ISO 7776 (X.75 SLP)
   Transparent
   LAPD with Q.921 for D channel X.25 (SAPI 16)
   T.30 for fax group 3
   ISO 7776 (X.75 SLP) with V.42bis compression
   V.120 asyncronous mode
   V.120 bit-transparent mode
B3 protocols support: 0x800000bf
   Transparent
   T.90NL, T.70NL, T.90
   ISO 8208 (X.25 DTE-DTE)
   X.25 DCE
   T.30 for fax group 3
   T.30 for fax group 3 with extensions
   Modem

  0100
  0200
  39000000
  1f010040
  1b0b0000
  bf000080
  00000000 00000000 00000000 00000000 00000000 00000000
  01000001 00020000 00000000 00000000 00000000

Supplementary services support: 0x000003ff
   Hold / Retrieve
   Terminal Portability
   ECT
   3PTY
   Call Forwarding
   Call Deflection
   MCID
   CCBS

/etc/asterisk/capi.conf:

Code:
[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

;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.


[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=DID     ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
                 ;when using NT-mode, 'DID' should be set in any case
msn = 878788XX
incomingmsn=*    ;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=1     ;capi controller number of this interface/port
group=1          ;dialout group

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=g1   ;context for incoming calls
holdtype=local    ;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. Disable this before you start recording voicemail
                 ;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)

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)

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.

[ISDN2]          ;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=DID     ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
                 ;when using NT-mode, 'DID' should be set in any case
msn=878788XX
incomingmsn=*    ;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=2     ;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=g1   ;context for incoming calls
holdtype=local    ;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. Disable this before you start recording voicemail
                 ;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
;echocancel=yes  ;Dialogic Diva (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)
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 Diva only)
;allow=all       ;RTP codec selection (valid with Dialogic Diva 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


[ISDN3]          ;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=*    ;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=3     ;capi controller number of this interface/port
group=2          ;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=g2   ;context for incoming calls
holdtype=local    ;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. Disable this before you start recording voicemail
                 ;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
;echocancel=yes  ;Dialogic Diva (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)
transfergroup=2 ;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 Diva only)
;allow=all       ;RTP codec selection (valid with Dialogic Diva 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

[ISDN4]          ;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=*    ;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=4     ;capi controller number of this interface/port
group=2          ;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=g2   ;context for incoming calls
holdtype=local    ;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. Disable this before you start recording voicemail
                 ;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)
;echocancel=yes  ;Dialogic Diva (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)
transfergroup=2 ;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 Diva only)
;allow=all       ;RTP codec selection (valid with Dialogic Diva 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

Habe übrigens für ISDN3 und 4 auch versucht mit isdnmode = DID. Leider ohne Erfolg.

CLI zeigt:
Code:
CAPI B-channel information:
Line-Name       NTmode state i/o bproto isdnstate   ton  number
----------------------------------------------------------------
ISDN4#02         yes   -----  -  trans              0x00 ''->''
ISDN4#01         yes   -----  -  trans              0x00 ''->''
ISDN3#02         yes   -----  -  trans              0x00 ''->''
ISDN3#01         yes   -----  -  trans              0x00 ''->''
ISDN2#02         no    -----  -  trans              0x00 ''->''
ISDN2#01         no    -----  -  trans              0x00 ''->''
ISDN1#02         no    Disc   -  trans              0x00 ''->''
ISDN1#01         no    -----  -  trans              0x00 ''->''

so, jetzt steh ich auf dem Schlauch und weiß nicht weiter. Wo liegt mein Fehler? kann mir bitte jemand einen Tip geben?
Danke!

Gruß
segelfreak
 
Das was du suchst/brauchst nennt sich NT Mode.
Das was AVM kann ist nur TE Mode. Sorry.
Du brauchst dafür andere Hardware.
(Beronet/Junghanns/Sirrix/Eicon)
 
avm kein NT Mode

Hallo und vielen Dank für die Antwort, auch wenn es genau das ist, was ich gar nicht hören wollte :mad:
Also dann werd ich mal los und mir eine HFC Karte besorgen.....(müssen)

Na ja, ist ja auch schön, mal zwei Ports in Reserve zu haben :D

Gruß
segelfreak
 
Abhänbgig davon *was* Dein Asterisk hinter der Hicom machen soll: für den professionellen Einsatz rate ich dringend davon ab, das mit ner AVM und ner HFC-basierten Karte zusammenzubasteln.

Entweder gleich richtig mit Karte von Dialogics, oder mit nem externen Mediengateway wie z.B. ner Patton SmartNode 4638.
 
Asterisk vor Hicom

Hallo foschi,

also hier get es in erster Linie um eine Minimallösung.
die Hicom soll zukünftig hinter dem * hängen, da es sich bei der anlage um eine getagtes Modell hadelt und die Möglichkeiten (Ansage, Musik, etc) so reichlich beschränkt sind.
Bisher hatte ich den * hinter der Hicom. Dies lief auch soweit einwandfrei mit iaxmodem für fax und zwei SIP Telefonen dran.

Nun soll der * am Anlagenanschluss den Vorreiter spielen und die Anrufe nach festen zeiten auf die SIP Tels, die Hicom und Mobilnummern verteilen.

Server ist ein Xeon 3Ghz, sollte also mit der Last einer, mglw. zweier HFC Karten sicher auskommen. Leider drängt auch ein wenig die Zeit bei mir - urlaub steht vor der Tür und nichts funzt jetzt richtig.

Dazu kommen blöde Probleme bei der Installation der HFC unter Ubuntu 9.04 (jaunty) offenbar passen die zaptel Treiber nicht mehr mit dem 28'er Kernel zusammen. Aber dazu poste ich gleihc mal im entsprechend subchannel. ich hoffe, dort kann manmir schnell helfen - bevor ich in die Klapse gehe :confused:

Gruss
segelfreak
 
Mit CPU-Last hat das ganze nichts zu tun.
Nimm ein Mediengateway - dann hört das Gebastel auf (dürfte auch wirtschaftlicher sein).
 
patton smartnode

Hallo Foschi,

ehrlich gesagt, finde ich den Preis für so ein Teil mit nur einem internen S0 ziemlich happig.
Wenn ich die Hicom mit nur einem Anschluß daran klemmen kann, ist der Spass damit auch schnell am Ende. Das reicht also wirklich nur gerade, um noch ein Fax daran zu betreiben.

Ich werd's jetzt wohl anders machen:

Ich erweitere die Hicom für 36 EUR um weitere 4 S0 ports.
Dann kommt der * wieder hinter die Hicom und die Anrufe werden komplett auf den * durchgeroutet. Wenn dann Nebenstellen von der Hicom angewählt werden müssen, dann reichen die 8 Kanäle insgesamt sicher aus.

Auf diese Weise erspare ich mir das gefrickel, denn die C4 läuft ja bereits ohne Probleme und die Installation war - für einen Anfänger wie mich - richtig einfach.

Gruß
segelfreak
 
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