Grundsätzliche Einstellungen an einem Testserver

[ueckerle]

Hallo,

ich habe vor hinter einer DeTeWE-Telefonanlage einen kleinen Asteriskserver zu betreiben.

Es sollen Möglichkeiten geschaffen werden um

Anrufe von der Telefonanlage und von Extern (über die Telefonanlage) an meinen Server gehen und das alle IP-Telefone nach draussen telfonieren können.

Um dieses Beispiel umsetzen zu können habe ich einen kleinen Server aufgesetzt mit folgenden Inhalt:

FS-PC mit 1 GB Ram,
HF-Cologne-Karte,
Elastix-Asterisk (mit CentOS)
und einigen Softphone und einem SPA942.

Folgende Asterisk-Version ist installiert: 1.4.25.1
mISDN: 1.1.9.1

Lasse ich misdn ohne Asterisk laufen kommt bei misdnportinfo folgendes:

[root@elastix ~]# misdnportinfo

Port  1: TE-mode BRI S/T interface line (for phone lines)
 -> Interface is Poin-To-Point.
 -> Protocol: DSS1 (Euro ISDN)
 -> childcnt: 2
--------

mISDN_close: fid(3) isize(131072) inbuf(0x80ad060) irp(0x80ad060) iend(0x80ad060)

Läuft ausserdem der Asterisk, sieht dies so aus:

[root@elastix ~]# misdnportinfo

Port  1: TE-mode BRI S/T interface line (for phone lines)
 -> Interface is Poin-To-Point.
 -> Protocol: DSS1 (Euro ISDN)
 -> Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib.
 -> childcnt: 2
 * Port NOT useable for PBX (maybe there is already a PBX running?)
--------

mISDN_close: fid(3) isize(131072) inbuf(0x8aab060) irp(0x8aab060) iend(0x8aab060)

Wo ist nur mein Fehler?

Denke ich falsch?

Mit freundlichen Grüßen

Ulrich

 

[ueckerle]

Meine CONF-Files

/etc/mISDN.conf:

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module: hfcsusb
Options: debug=<number> poll=<number>
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module: xhfc
Options: debug=<number>
Port Attributes: mode=(te|nt), link=(ptp|ptmp), line=(s0|up) capi=(yes|no)
                 lineloop_b1=(yes|no) lineloop_b2=(yes|no), lineloop_d=(yes|no)
                 polx=(yes|no)
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module: mISDN_dsp
Options: debug=<number>, options=<number>, poll=<number>,
         dtmfthreshold=<number>
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-->
<mISDNconf>
        <module poll="128" debug="0" timer="no">hfcmulti</module>
        <module debug="0" options="0">mISDN_dsp</module>
        <devnode user="asterisk" group="asterisk" mode="660">mISDN</devnode>
        <card type="hfcpci">
                <port mode="te" link="ptp">1</port>
        </card>
</mISDNconf>

/etc/misdn-init.conf:

# dsp_options=<number>
#
#       set this to 2 and you'll have software bridging instead of
#       hardware bridging.
#
#
# dtmfthreshold=<milliseconds>
#
#       Here you can tune the sensitivity of the dtmf tone recognizer.
#
# timer=<1|0>
#
#       set this to 1 if you want hfcmulti to register at ztdummy (zaptel)
#       and provide a 1khz timing source for it. This makes it possible
#       to have an accurate timing source for asterisk through zaptel from
#       hfcmulti to make applications like Meetme and faxing between wctdm
#       and hfcmulti work properly.
#
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0

/etc/asterisk/misdn.conf:

...
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 either value the presentation indicators are used
; from asterisks SetCallerPres application.
; s=0, p=0 -> callerid presented
; s=1, p=1 -> callerid restricted (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=no

; 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

 

[JanG80]

Wo genau ist dein Problem? misdnportinfo sagt dir doch nur, dass Asterisk (der PBX) bereits läuft. Soweit also alles richtig gemacht.

 

[ueckerle]

Welche Einstellungen in Asterisk muß ich den jetzt machen, damit ich den PC/Telefon anrufen kann?

Grüsse

Ulrich

 

[JanG80]

Es macht keinen Sinn, das jetzt hier komplett durchzukauen. Du solltest dich gründlich einlesen, wenn du mit Asterisk arbeiten willsz. Im Internet finden sich unzählige Howtos zu dem Thema. Stichwort Dialplan bzw. extensions.conf. Wenn du dich eingelesen hast, und sich dann Fragen zu den Details stellen, kannst du dich ja wieder hiermelden.