misdn.conf erklären

frindly

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Hallo
kann mir jemand kurz Beschreiben, was die Einstellungen in meiner misdn.conf bedeuten?


Code:
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/asterisk/misdn-nt.log

[default]
language=de
musicclass=default
internationalprefix = 00
nationalprefix = 0
max_incoming=-1
max_outgoing=-1
senddtmf=yes
reject_cause=17

[misdn1]
ports=1,2,3,4,5,6,7,8,9
context=incoming
echocancel=yes

[misdn2]
ports=16
context=_mh1
overlapdial=yes
dialplan = 4
localdialplan = 4

[misdn9]
ports=16
context=_mh1
overlapdial=yes
dialplan = 4
localdialplan = 4
 
Hallo frindly,

hier mal eine misdn.conf.sample, wo vieles beschrieben ist.

Svenja

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=no


;
; 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=0

; 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.
;

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                              ; SIP channel. Defaults to "no". An enabled jitterbuffer will
                              ; be used only if the sending side can create and the receiving
                              ; side can not accept jitter. The SIP channel can accept jitter,
                              ; thus a jitterbuffer on the receive SIP side will be used only
                              ; if it is forced and enabled.

; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
                              ; channel. Defaults to "no".

; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                              ; resynchronized. Useful to improve the quality of the voice, with
                              ; big jumps in/broken timestamps, usually sent from exotic devices
                              ; and programs. Defaults to 1000.

; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
                              ; channel. Two implementations are currently available - "fixed"
                              ; (with size always equals to jbmaxsize) and "adaptive" (with
                              ; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[default]

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

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

;
; 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 list which bearer capabilities should be allowed:
;   all                  - allow any bearer capability
;   speech               - allow speech
;   3_1khz               - allow 3.1KHz audio
;   digital_unrestricted - allow unrestricted digital
;   digital_restricted   - allow restricted digital
;   video                - allow video
;
; Example:
; allowed_bearers=speech,3_1khz
;
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=0

; 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=no

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

; 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=no

;
; 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=-1

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

[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order) 
ports=1,2
; context where to go to when incoming Call on one of the above ports
context=Intern

;[internPP]
;
; 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=3
	
;[first_extern]
; again port defs
;ports=4
; again a context for incoming calls
;context=Extern1
; msns for te ports, listen on those numbers on the above ports, and 
; indicate the incoming calls to asterisk
; here you can give a comma separated list or simply an '*' for 
; any msn. 
;msns=*

; here an example with given msns
;[second_extern]
;ports=5
;context=Extern2
;callerid=15
;msns=102,144,101,104
 
Bei der Gelegenheit fällt mir ein, dass ich schon immer mal zur Diskussion stellen wollte, welchen Grund es geben könnte, bridging auf "no" zu stellen.
 
Ich nehme einfach mal "HFC-Karte" als Ausrede. Aber ehrlich gesagt :noidea:

Svenja
 
ich hab irgendwo gelesen, das mit bridgin der kernel abstürzt.
ohne jetzt genau zu wissen, in welchen zusammenhang und warum.
:-Ö
 
wenn ich debu auf 2 setzte,
dann bekomm ich ja mehr infos was alles passiert.
landet das alles in der misdn.log ???
 
ich hab irgendwo gelesen, das mit bridgin der kernel abstürzt.

Das kann ich jetzt zumindest auswendig nicht bestätigen, hab das immer auf yes (mit HFC basierten Karten) zumal mir das von der Doku her als sinnvoll erscheint.
 
diese Meldung häuft sich in der misdn.log: (noch ohne höheren debug level):

Mon Jun 7 10:46:39 2010: P[ 3] --> not yet handled: facility type:0xffff


was heisst das?
 
in meiner datei seh ich
misdn 1
misdn 2
misdn 9
warum?
ich hab ja nur zwei beronet karten, jetweils mit 8 ports.
ist die aufteilung sinnvoll?
 
Warum die bei Dir so eingetragen sind, kann ich dir nicht sagen. Aber das sind ja in dem Fall nur die Namen der Portgruppen. So eine Gruppe könnte genauso gut Susi heißen (du müsstest sie halt dann beim Dial mit Susi ansprechen). Von daher keine Fehlerquelle in diesem Sinne.
 
ja. dann sind das gruppennamen.
aber was bedeutet in misdn1 das dort die ports bis 8 aufgeführt sind,
einzelln,
während bei misdn2 und misdn 9 nur ports 16 steht?
was ist mit den ports dazwischen?
 
Das wären gute Fragen an denjenigen, der diese misdn.conf geschrieben hat.
 
Hallo zusammen,

kann mir jemand sagen was passieren könnte wenn ich in dialplan options von 0 auf 2 ändere ?

gruß
kiru
 

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