FRITZ!Card + Trixbox = jede menge graue Haare

[Scom]

hallo,

hier erstmal die ausgangslage:

- trixbox 2.2 (asterisk 1.2.20, centos 4.5 kernel 2.6.9-42.0.10.EL)
- chan_capi_cm 0.6.5_1.2.8-1
- fritz!card pci v2.0 isdn (rev 01) (hängt direkt an der ntba)
- 2x elmeg ip290 sip telefone

ziel:
- raus/rein telefonieren über isdn leitung

problem:

- ich bin mit dem client verbunden (asterisk -rvvv) und wähle über das telefon eine nummer nach draußen, dann fängt er an und kein ton kommt beim telefon, dazu kommt noch das der client von asterisk mich automatisch disconnected
- ich habe in der trixbox eingestellt das er bei ausgehenden routen ein "r" dran hängen soll für ein ringing. das tut er auch und wenn man raus telefoniert klingelt das telefon (hab mein handy zum testen genommen).. wenn ich dann aber abnehme, legt mein handy automatisch auf und das telefon klingelt weiter, dazu kommt wieder das der client sich automatisch disconnected
- rein telefonieren geht gleich mal gar nicht, da zeigt der client an das er angerufen wird, aber dann disconnected der client wieder und nix passiert


hier mal ein paar configs:

hier das beispiel wo ich vom internen sip telefon über isdn auf mein handy telefonieren will (nummer ist von mir unkenntlich gemacht wurden), dort wo answered kommt, da habe ich abgenommen, danach bricht er ab.

[root@asterisk1 ~]# asterisk -rvvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.20, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.20 currently running on asterisk1 (pid = 4921)
Verbosity is at least 4
    -- Executing Macro("SIP/2001-088b10c0", "dialout-trunk|2|017612312312||") in new stack
    -- Executing Set("SIP/2001-088b10c0", "DIAL_TRUNK=2") in new stack
    -- Executing Set("SIP/2001-088b10c0", "_NODEST=") in new stack
    -- Executing Set("SIP/2001-088b10c0", "DIAL_NUMBER=017612312312") in new stack
    -- Executing Set("SIP/2001-088b10c0", "ROUTE_PASSWD=") in new stack
    -- Executing Set("SIP/2001-088b10c0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?noauth") in new stack
    -- Goto (macro-dialout-trunk,s,8)
    -- Executing Set("SIP/2001-088b10c0", "GROUP()=OUT_2") in new stack
    -- Executing Macro("SIP/2001-088b10c0", "user-callerid|SKIPTTL") in new stack
    -- Executing NoOp("SIP/2001-088b10c0", "user-callerid: device 2001") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?report") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?start") in new stack
    -- Executing Set("SIP/2001-088b10c0", "REALCALLERIDNUM=2001") in new stack
    -- Executing NoOp("SIP/2001-088b10c0", "REALCALLERIDNUM is 2001") in new stack
    -- Executing Set("SIP/2001-088b10c0", "AMPUSER=2001") in new stack
    -- Executing Set("SIP/2001-088b10c0", "AMPUSERCIDNAME=Sven") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?report") in new stack
    -- Executing Set("SIP/2001-088b10c0", "CALLERID(all)=Sven <2001>") in new stack
    -- Executing Set("SIP/2001-088b10c0", "REALCALLERIDNUM=2001") in new stack
    -- Executing NoOp("SIP/2001-088b10c0", "TTL:  ARG1: SKIPTTL") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/2001-088b10c0", "Using CallerID "Sven" <2001>") in new stack
    -- Executing Macro("SIP/2001-088b10c0", "record-enable|2001|OUT") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/2001-088b10c0", "recordingcheck|20070703-091641|1183447001.0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070703-091641|1183447001.0: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/2001-088b10c0", "No recording needed") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?skipoutcid") in new stack
    -- Executing Set("SIP/2001-088b10c0", "DIAL_TRUNK_OPTIONS=r") in new stack
    -- Executing Macro("SIP/2001-088b10c0", "outbound-callerid|2") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/2001-088b10c0", "REALCALLERIDNUM is 2001") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing Set("SIP/2001-088b10c0", "USEROUTCID=") in new stack
    -- Executing Set("SIP/2001-088b10c0", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/2001-088b10c0", "TRUNKOUTCID=5") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing GotoIf("SIP/2001-088b10c0", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,18)
    -- Executing GotoIf("SIP/2001-088b10c0", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing NoOp("SIP/2001-088b10c0", "CallerID set to "Sven" <2001>") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?nomax") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "0?chanfull") in new stack
    -- Executing DeadAGI("SIP/2001-088b10c0", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/2001-088b10c0", "OUTNUM=017612312312") in new stack
    -- Executing Set("SIP/2001-088b10c0", "custom=AMP") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?customtrunk") in new stack
    -- Goto (macro-dialout-trunk,s,22)
    -- Executing Set("SIP/2001-088b10c0", "pre_num=AMP:CAPI/ISDN1/") in new stack
    -- Executing Set("SIP/2001-088b10c0", "the_num=OUTNUM") in new stack
    -- Executing Set("SIP/2001-088b10c0", "post_num=") in new stack
    -- Executing GotoIf("SIP/2001-088b10c0", "1?outnum:skipoutnum") in new stack
    -- Goto (macro-dialout-trunk,s,26)
    -- Executing Set("SIP/2001-088b10c0", "the_num=017612312312") in new stack
    -- Executing Dial("SIP/2001-088b10c0", "CAPI/ISDN1/017612312312|300|r") in new stack
    -- Called ISDN1/017612312312
    -- CAPI/ISDN1/017612312312-0 is ringing
    -- CAPI/ISDN1/017612312312-0 answered SIP/2001-088b10c0
asterisk1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@asterisk1 ~]#

hier das beispiel für einen eingehenden anruf (nummer sind wieder unkenntlich gemacht):

[root@asterisk1 ~]# asterisk -rvvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.20, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.20 currently running on asterisk1 (pid = 5241)
Verbosity was 1 and is now 4
  == ISDN1: Incoming call '017612312312' -> '3333333'
asterisk1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@asterisk1 ~]#

hier die capi.conf:

;
; CAPI config
;
;

; 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
;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=*    ;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=on         ;enable use of Q.SIG extensions.

die extensions_custom.conf:

; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP's Asterisk dialplan.

; All custom context should contain the string 'custom' in it's name 

; Extensions in AMP have access to the 'from-internal' context.
; The context 'from-internal-custom' is included in 'from-internal' by default

#include extensions_trixbox.conf
#include extensions_hud.conf

[from-internal-custom]

include => from-internal-trixbox
			
;1234,1,Playback(demo-congrats)		; extensions can dial 1234
;1234,2,Hangup()
;h,1,Hangup()
;include => custom-recordme			; extensions can also dial 5678

; custom-count2four,s,1 can be used as a custom target for
; a Digital Receptionist menu or a Call Group
;[custom-count2four]		
;s,1,SayDigits(1234)
;s,2,Hangup

; custom-recordme,5678,1 can be used as a custom target for
; a Digital Receptionist menu or a Call Group
;[custom-recordme]
;exten => 5678,1,Wait(2)
;exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
;exten => 5678,3,Wait(2)
;exten => 5678,4,Playback(/tmp/asterisk-recording)
;exten => 5678,5,Wait(2)
;exten => 5678,6,Hangup 

[capi-in]
include => from-pstn

capiinfo gibt folgendes aus:

[root@asterisk1 ~]# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07  (49.23)
Serial Number: 1000001
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

der trunkdialstring ist:

CAPI/ISDN1/$OUTNUM$

wenn noch was fehlt, bitte bescheid sagen..

hoffe jemand kann helfen =)

 

[achim22]

Hi!
Kann Dir zwar im Augenblick nicht bei Deinem Problem helfen, aber kannst Du mir bitte verraten wie die die Fritzcard eingebunden bekommen hast?

Ich (muss) mich auch gerade mit der Trixbox rumärgern.

Unter lspci erhalte ich folgendes:

Network controller: AVM Audiovisuelles MKTG & Computer System GmbH Fritz!PCI v2.0 ISDN (rev 02)

Unter Packages kann ich den chan_capi-cm.i386 installieren. Wenn ich diesen jedoch in der modules.con mit load => chan_capi.so eintrage sagt mir asterisk das er das modul nicht findet.
Auch den Eintrag unter global .....chan_capi.so=yes habe ich gemacht.

irgendwie habe ich auch ein Problem mit capiinfo........capi not installed

 

[MarcelDeLuX]

Versuch doch einfach mal innerhalb des Context capi-in nen Dialplan auf dein Telefon zumachen, anschließend nen reload und nen anruf von deinem Handy:

exten => 3333333,1,Dial(SIP/2001)