mISDN config ?!?!

foexle

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Hallo Leute,
ich habe hier 2 Bücher liegen und ettliche HowTo's zu der installation von asterisk und mISDN.
Installiert ist soweit auch alles, nur ich komme einfach mit der Konfiguration nicht klar.

bei
Code:
sudo /etc/init.d/mISDN-init scan
findet er CARD=1,avmfritz

nach config steht das auch so in der /etc/misdn-init.config

wenn ich nun
in dieser misdn-init.config

option=1,master-clock anstelle (zum Sync mit amt) will den daemon restarten, stürzt mein gesamtes Debian system ab und kann es nicht mehr hochfahren ... erst nachdem ich wieder die misdn gelöscht habe ... es nutzt noch nicht mal wenn ich wieder diese option auskommentiere.

Das weitere ist, das ich unter misdnportinfo

Code:
sudo misdnportinfo

Found no card. Please be sure to load card drivers.

mISDN_close: fid(3) isize(131072) inbuf(0x804c060) irp(0x804c060) iend(0x804c060)
das bekomme .... mit sudo modprobe avmfritz, wird das module zwar geladen, taucht aber imme rnoch nicht in der info auf ...

was mache ich falsch ?
oder habe ich was vergessen zu installieren ?


greetz
 
Ist mISDN denn überhaupt gestartet?

Code:
/etc/init.d/misdn-init start
 
huhu
und danke für wenichstens eine Antwort :>

ja es wird gestartet

Code:
 sudo /usr/sbin/misdn-init start
-----------------------------------------
 Loading module(s) for your misdn-cards:
-----------------------------------------
/sbin/modprobe --ignore-install avmfritz protocol=0x2 layermask=0xf 
/sbin/modprobe mISDN_dsp debug=0x0 options=0 poll=128 dtmfthreshold=100
 
wuuussaaaa super :> so ein howtoo habsch gesucht :D

ok ich werde das mal durchackern vielen dank !!!!!!
 
so das hat alles super geklappt .... sher schönes howto !
also asterisk und mISDN Interface läuft :> soweit ich das beurteilen kann.

Ich habe nun mal den rechner an das Telefonnetz der firma angeschlossen.
Die nummer wurde gapatcht. Wie kann ich nun überprüfen, ob anrufe eingehen bei meinem Server, oder auf der Schnittstelle?
Vorallem wie gehts nun weiter ? ... Als nächstes ist wohl die richtige Konfiguration der mISDN dran und der Wählplan oder ?
Oder habe ich was vergessen was noch zuvor gemacht werden muss ?


Greetz

foexle



edit: ich habe nun mal in der Asterisk cli mich bissal umgeschaut und habe geade
mit
misdn show ports stats

kommt da nur out_calls 0
wiso kommt da kein in_calls? hab ich da was falsch eingestellt ? ... weil ich brauchs genau anderst rum ;) der server muss nur anrufe entgegen nehmen ;)


edit2 :
Wenn ich meinen asterisk anwähle, kommt der fehler in der CLI
Code:
 WARNING[3365]: chan_misdn.c:4336 cb_events: Extension can never match, so disconnecting on port(1).maybe you want to add an 'i' extension to catch this case.

meine misdn.conf schaut so aus

Code:
;
; chan_misdn sample config
;

; general section:
;
; for debugging and general setup, things that are not bound to port groups
;

[general] 
;
; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
;
misdn_init=/etc/misdn-init.conf

; set debugging flag: 
;   0 - No Debug
;   1 - mISDN Messages and * - Messages, and * - State changes
;   2 - Messages + Message specific Informations (e.g. bearer capability)
;   3 - very Verbose, the above + lots of Driver specific infos
;   4 - even more Verbose than 3
;
; default value: 0
;
debug=0



; set debugging file and flags for mISDNuser (NT-Stack) 
; 
; flags can be or'ed with the following values:
;
; DBGM_NET        0x00000001
; DBGM_MSG        0x00000002
; DBGM_FSM        0x00000004
; DBGM_TEI        0x00000010
; DBGM_L2         0x00000020
; DBGM_L3         0x00000040
; DBGM_L3DATA     0x00000080
; DBGM_BC         0x00000100
; DBGM_TONE       0x00000200
; DBGM_BCDATA     0x00000400
; DBGM_MAN        0x00001000
; DBGM_APPL       0x00002000
; DBGM_ISDN       0x00004000
; DBGM_SOCK       0x00010000
; DBGM_CONN       0x00020000
; DBGM_CDATA      0x00040000
; DBGM_DDATA      0x00080000
; DBGM_SOUND      0x00100000
; DBGM_SDATA      0x00200000
; DBGM_TOPLEVEL   0x40000000
; DBGM_ALL        0xffffffff
;

ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log


; some pbx systems do cut the L1 for some milliseconds, to avoid 
; dropping running calls, we can set this flag to yes and tell
; mISDNuser not to drop the calls on L2_RELEASE
ntkeepcalls=no

; the big trace
;
; default value: [not set]
;
;tracefile=/var/log/asterisk/misdn.log


; set to yes if you want mISDN_dsp to bridge the calls in HW
;
; default value: yes
;
bridging=yes


;
; watches the L1s of every port. If one l1 is down it tries to 
; get it up. The timeout is given in seconds. with 0 as value it
; does not watch the l1 at all
; 
; default value: 0
;
; this option is only read at loading time of chan_misdn, 
; which means you need to unload and load chan_misdn to change the 
; value, an asterisk restart should do the trick
; 
l1watcher_timeout=20

; stops dialtone after getting first digit on nt Port
;
; default value: yes
;
stop_tone_after_first_digit=yes

; whether to append overlapdialed Digits to Extension or not 
;
; default value: yes
;
append_digits2exten=yes

;;; CRYPTION STUFF

; Whether to look for dynamic crypting attempt
;
; default value: no
;
dynamic_crypt=no

; crypt_prefix, what is used for crypting Protocol
;
; default value: [not set]
;
crypt_prefix=**

; Keys for cryption, you reference them in the dialplan
; later also in dynamic encr.
;
; default value: [not set]
;
crypt_keys=test,muh

; users sections:
; 
; name your sections as you which but not "general" ! 
; the sections are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name, 
; chan_misdn tries every port in this section to find a 
; new free channel
; 

; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;

[default]

; define your default context here
;
; default value: default
;
context=misdn

; language
;
; default value: en
;
language=de

;
; sets the musiconhold class
;
musicclass=default

;
; Either if we should produce DTMF Tones ourselves
; 
senddtmf=yes

;
; If we should generate Ringing for chan_sip and others
;
far_alerting=no


;
; here you can define which bearers should be allowed
;
allowed_bearers=all

; Prefixes for national and international, those are put before the 
; oad if an according dialplan is set by the other end. 
;
; default values: nationalprefix      : 0
;                 internationalprefix : 00
;
nationalprefix=0
internationalprefix=00

; set rx/tx gains between -8 and 8 to change the RX/TX Gain
;
; default values: rxgain: 0
;                 txgain: 0
;
rxgain=0
txgain=-1

; some telcos especially in NL seem to need this set to yes, also in 
; switzerland this seems to be important
;
; default value: no
;
te_choose_channel=no



;
; This option defines, if chan_misdn should check the L1 on  a PMP 
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
; because of a lost Link or because the Provider shut it down...
;
; default: no
;
pmp_l1_check=no


;
; in PMP this option defines which cause should be sent out to 
; the 3. caller. chan_misdn does not support callwaiting on TE
; PMP side. This allows to modify the RELEASE_COMPLETE cause 
; at least.
;
reject_cause=16


;
; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING), 
; this requests additional Infos, so we can waitfordigits 
; without much issues. This works only for PTP Ports
; 
; default value: no
;
need_more_infos=no


;
; set this to yes if you want to disconnect calls when a timeout occurs
; for example during the overlapdial phase
;
nttimeout=no

; set the method to use for channel selection:
;   standard    - always choose the first free channel with the lowest number
;   round_robin - use the round robin algorithm to select a channel. use this
;                 if you want to balance your load.
;
; default value: standard
;
method=standard


; specify if chan_misdn should collect digits before going into the 
; dialplan, you can choose yes=4 Seconds, no, or specify the amount
; of seconds you need;
; 
overlapdial=yes

;
; dialplan means Type Of Number in ISDN Terms (for outgoing calls)
;
; there are different types of the dialplan:
;
; dialplan -> outgoing Number
; localdialplan -> callerid
; cpndialplan -> connected party number
;
; dialplan options: 
;
; 0 - unknown
; 1 - International
; 2 - National
; 4 - Subscriber
;
; This setting is used for outgoing calls
;
; default value: 0
;
dialplan=0
localdialplan=0
cpndialplan=0



;
; turn this to no if you don't mind correct handling of Progress Indicators  
;
early_bconnect=yes


;
; turn this on if you like to send Tone Indications to a Incoming
; isdn channel on a TE Port. Rarely used, only if the Telco allows
; you to send indications by yourself, normally the Telco sends the 
; indications to the remote party.
; 
; default: no
;
incoming_early_audio=yes

; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
; isdn overlap dial. 
; note: This will jump into the s exten for every exten!
;
; default value: no
;
;always_immediate=no

;
; set this to yes if you want to generate your own dialtone 
; with always_immediate=yes, else chan_misdn generates the dialtone
;
; default value: no
;
nodialtone=no


; uncomment the following if you want callers which called exactly the 
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension 
; instead
;
; default value: no
;
;immediate=no

; uncomment the following to have hold and retrieve support
;
; default value: no
;
;hold_allowed=yes

; Pickup and Callgroup
;
; default values: not set = 0
; range: 0-63
;
;callgroup=1
;pickupgroup=1


;
; these are the exact isdn screening and presentation indicators
; if -1 is given for both values the presentation indicators are used
; from asterisks SetCallerPres application.
; s=0, p=0 -> callerid presented not screened
; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
; 
; default values s=-1, p=-1
presentation=-1
screen=-1

; this enables echocancellation, with the given number of taps
; be aware, move this setting only to outgoing portgroups!
; A value of zero turns echocancellation off.
;
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
;
; default value: no
;
echocancel=256

; Set this to no to disable echotraining. You can enter a number > 10
; the value is a multiple of 0.125 ms. 
;
; default value: no 
; yes = 2000
; no = 0
;
echotraining=2000

;
; chan_misdns jitterbuffer, default 4000
; 
jitterbuffer=4000

;
; change this threshold to enable dejitter functionality
;
jitterbuffer_upper_threshold=0


;
; change this to yes, if you want to bridge a mISDN data channel to 
; another channel type or to an application.
;
hdlc=no


;
; defines the maximum amount of incoming calls per port for
; this group. Calls which exceed the maximum will be marked with 
; the channel variable MAX_OVERFLOW. It will contain the amount of 
; overflowed calls
;
max_incoming=4

;
; defines the maximum amount of outgoing calls per port for this group
; exceeding calls will be rejected
;
max_outgoing=-1


[intern]
;
; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
; configs. For backwards compatibility you can still set ptp here.
ports=1
context=isdn_in
msns=*


wo könnte der fehler liegen ?
 
Zuletzt bearbeitet:
Der Fehler dürfte darin liegen, dass dein Dialplan (extensions.ael oder extensions.conf) keine passende Extension im Kontext isdn_in enthält.
 
also muss ihc nur in der extension.conf

[isdn_in]
extern => ...

eintragen?

werde das gleich mal versuchen !!!

danke schon mal für die hilfe


ich hab gerade mal in der misdn.log nachgeschaut da steht nur

Code:
Tue Aug  5 15:54:32 2008: P[ 1]  channel with stid:0 for one second still in use!
Tue Aug  5 16:30:01 2008: P[ 0]  -- mISDN Channel Driver Registered --
Tue Aug  5 16:35:40 2008: P[ 1]  channel with stid:0 for one second still in use!
Tue Aug  5 16:35:44 2008: P[ 1]  channel with stid:0 for one second still in use!
Tue Aug  5 16:38:04 2008: P[ 1]  channel with stid:0 for one second still in use!
Tue Aug  5 16:38:18 2008: P[ 1]  channel with stid:0 for one second still in use!
Tue Aug  5 16:41:02 2008: P[ 1]  channel with stid:0 for one second still in use

das hier drin ... sollte da nicht bischen mehr drin stehen, wenn ein anruf ankommt ?
 
Zuletzt bearbeitet:
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