Anrufen nicht möglich

[GregorFranz]

Hallo!

Ich bin Linux Anfänger!
(Anlagenanschluss Österreich)

Benutzt habe ich folgende Anleitungen:

http://pbx-manager.de/installation-linux-asterisk.php

http://www.tecchannel.de/telko/sprache/432904/
http://www.tecchannel.de/telko/sprache/434680/
http://www.tecchannel.de/telko/sprache/434766/


Ich habe ein X-Lite Softphone auf meinem PC installiert und kann damit über den Suse 10.0 Server mit PCIFritzCard raustelefonieren.
Wenn ich mich aber selbst anrufen will, bekomme ich immer nur ein Besetzt und am Asterisk Server selbst kommt nicht mal irgendeine Meldung!!

Woran könnte das liegen?

Danke!

MfG Gregor

 

[armincm]

Ohne deiner Configs (capi.conf, sip.conf, extensions.conf, ...) kann man da leider nicht so viel sagen.

Armin

 

[GregorFranz]

Hallo!

Hier sind meine configs!









capi.conf:

;
; CAPI config
;
;

; general section

[general]
nationalprefix=1
internationalprefix=0043
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law

; interface sections ...

[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;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
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 to use
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
accountcode= ;Asterisk accountcode to use in CDRs
context=from-outside ;context for incoming calls
;holdtype=hold ;when Asterisk puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and Asterisk 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
;(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
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;language=de ;set language for this device (overwrites default language)
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)








sip.conf:

[general]
port=5060
bindaddr=0.0.0.0
context=sip-in
qualify=no
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes
regcontext=local-sip
register=1234567:[email protected]/001

[Username]
type=friend
username=Username
secret=Passwort
host=dynamic
context=local-sip
regexten=666

[sip-out]
type=peer
insecure=yes
nat=yes
username=1234567
fromuser=1234567
fromdomain=sipgate.at
secret=Passwort
host=sipgate.at
qualify=yes







sip.conf.old:

[general]
port=5060
bindaddr=0.0.0.0
context=sip-in
qualify=no
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes
regcontext=local-sip
register=1234567:[email protected]/001

[Username]
type=friend
username=Username
secret=Passwort
host=dynamic
context=local-sip
regexten=666








extensions.conf:

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; Third Lane Technologies, LLC
; Asterisk PBX Manager
; Sample configuration file
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
static=yes
writeprotect=no

[globals]
OFFICE_OPEN_OVERRIDE=
#include trunks.include
DIALOUT=9
INTERNATIONAL-PREFIX=011
RINGTIME=15

[from-outside]

exten => _X.,1,Goto(from-outside-redir,${EXTEN},1)
exten => s,1,Goto(from-outside-redir,${EXTEN},1)

[from-outside-redir]
#include "inbound.include"
#include "inbound_actions.include"
#include extensions.include
#include outbound.include
#include scripts.include
#include menus.include

[from-inside]
; local extensions are in extensions.include file
include => local-extensions
include => feature-extensions
; outbound numbers are in outbound.include file
include => outgoing
;parked calls
include => parkedcalls

exten => t,1,Hangup
exten => h,1,Hangup
exten => i,1,Playback(invalid)
exten => i,2,Hangup

[default]
include => local-extensions

[tl-record-app]

exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,BackGround(tl/record-menu)

; Recording
exten => 1,1,Wait(1)
exten => 1,2,Read(ID|tl/record-enter-num)
exten => 1,3,Wait(2)
exten => 1,4,Record(ogm/${ID}:gsm)
exten => 1,5,Wait(2)
exten => 1,6,Playback(ogm/${ID})
exten => 1,7,Wait(1)
exten => 1,8,Goto(s|2)
; Review
exten => 2,1,Wait(1)
exten => 2,2,Read(ID|tl/play-enter-num)
exten => 2,3,Wait(1)
exten => 2,4,Playback(ogm/${ID})
exten => 2,5,Wait(1)
exten => 2,6,Goto(s|2)

exten => t,1,Hangup
exten => h,1,Hangup
exten => #,1,Hangup

exten => i,1,Playback(invalid)
exten => i,2,Goto(s|2)

[sip-in]
exten=001,1,Dial(SIP/Username)

[local-sip]
exten=_XXXX.,1,Dial(CAPI/contr1/${EXTEN},30)
exten=_XXXX.,2,Hangup

[capi-in]
exten=9876543,1,Dial(SIP/Username,30)
exten=t,1,Hangup







extensions.conf.old:

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; Third Lane Technologies, LLC
; Asterisk PBX Manager
; Sample configuration file
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
static=yes
writeprotect=no

[globals]
OFFICE_OPEN_OVERRIDE=
#include trunks.include
DIALOUT=9
INTERNATIONAL-PREFIX=011
RINGTIME=15

[from-outside]

exten => _X.,1,Goto(from-outside-redir,${EXTEN},1)
exten => s,1,Goto(from-outside-redir,${EXTEN},1)

[from-outside-redir]
#include "inbound.include"
#include "inbound_actions.include"
#include extensions.include
#include outbound.include
#include scripts.include
#include menus.include

[from-inside]
; local extensions are in extensions.include file
include => local-extensions
include => feature-extensions
; outbound numbers are in outbound.include file
include => outgoing
;parked calls
include => parkedcalls

exten => t,1,Hangup
exten => h,1,Hangup
exten => i,1,Playback(invalid)
exten => i,2,Hangup

[default]
include => local-extensions

[tl-record-app]

exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,BackGround(tl/record-menu)

; Recording
exten => 1,1,Wait(1)
exten => 1,2,Read(ID|tl/record-enter-num)
exten => 1,3,Wait(2)
exten => 1,4,Record(ogm/${ID}:gsm)
exten => 1,5,Wait(2)
exten => 1,6,Playback(ogm/${ID})
exten => 1,7,Wait(1)
exten => 1,8,Goto(s|2)
; Review
exten => 2,1,Wait(1)
exten => 2,2,Read(ID|tl/play-enter-num)
exten => 2,3,Wait(1)
exten => 2,4,Playback(ogm/${ID})
exten => 2,5,Wait(1)
exten => 2,6,Goto(s|2)

exten => t,1,Hangup
exten => h,1,Hangup
exten => #,1,Hangup

exten => i,1,Playback(invalid)
exten => i,2,Goto(s|2)

[sip-in]
exten=001,1,Dial(SIP/Username)







modules.conf:

;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; The 'modem' channel driver and its subdrivers are
; obsolete, don't load them.
;
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
;
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
;noload => chan_alsa.so
;noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
chan_capi.so=yes







modules.conf.old:

;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; The 'modem' channel driver and its subdrivers are
; obsolete, don't load them.
;
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
;
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
;noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]

[global]
chan_capi.so=yes






Danke!

MfG
Gregor

 

[caMpi]

Moin,

ich denke der Fehler liegt in deiner extensions.conf und zwar:

[local-sip]
exten=_XXXX.,1,Dial(CAPI/contr1/${EXTEN},30)
exten=_XXXX.,2,Hangup

[capi-in]
exten=9876543,1,Dial(SIP/Username,30)
exten=t,1,Hangup

Nimm dir mal die Zeile
exten=_XXXX.,1,Dial(CAPI/contr1/${EXTEN},30)
vor, und änderst contr1 in ISDN1 (wie es auch in deiner capi.conf steht) um.
Und unter capi-in nimmst du statt SIP/Username einfach SIP/Rufnummer
Danach sollte zumindest eingehend was passieren.

Hoffe das hilft dir weiter.

Achso: hängst du irgendwie mit deinem Asterisk noch hinter ner TK-Anlage?

 

[GregorFranz]

Hallo!

Leider geht immer noch nichts!


Ich hänge mit der FritzCard PCI (die im Asterisk Server steckt) direkt am NTBA (anlagenanschluss)

Ich kann zwar über ISDN jede Nummer anrufen (über Softphone auf XP Workstation) aber jeder eingehende Anruf an die Rufnummer des NTBA wo die Fritz Card hängt, ergibt den Ton für Nummer ungültig (diese drei ansteigenden Pieptöne)

Danke für jede Hilfe!

MfG
Gregor

 

[GregorFranz]

Hallo nochmal!

Ich muss mich korrigieren:

Wenn ich Asterisk mit: asterisk -vvvvvvgc starte kommt sofort die Meldung:

--parse_srv: SRV mapped to host sipgate.at, port 5060

diese wiederholt sich im Minutentakt

Wenn ich mit den jetzigen Einstellungen die ISDN Nummer anrufe auf der die FritzCard hängt, kommt am Asterisk Server die Meldung:

==ISDN1: Incoming call ´Rufnummer´->´´
==ISDN1: CAPI Hangup

Im Telefon mit dem ich anrufe höre ich alledings immer nur: tut tut tut, tut tut tut, usw.

 

[caMpi]

änder in der capi.conf mal den isdnmodus auf msn

 

[GregorFranz]

Ich versteh die Welt nicht mehr :noidea:

Wenn ich meine ISDN Nummer von irgendeinem Mobiltelefon anrufe, komme ich auf die Asterisk Computerstimme (statt auf das Softphone)

Wenn ich meine ISDN Nummer aber von irgendeinem Festnetztelefon anrufe, passiert nichts (ausser tut tut tut, tut tut tut, usw.)

Ich bin dankbar für jeden Ratschlag!

MfG
Gregor

 

[SC50]

Ich hänge mit der FritzCard PCI (die im Asterisk Server steckt) direkt am NTBA (anlagenanschluss)

Hmmm,
eine FritzCard am Anlagenanschluß zu betreiben ist glaub ich keine so gute Lösung. Soll mit mISDN wohl funktionieren, aber gebaut wurde die Karte dafür nicht.

Gruß