Asterisk liest extension.conf nicht ??

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Hallo,

ich habe mir vorgenommen einen gateway zwischen der lokalen ISDN Telefonanlage und einem SIP Provider zu realisieren.

In mühevoller kleinarbeit habe ich es auch hinbekommen auf einem OpenSUSE 10 capi 2.0 und Asterisk aktuelle CVS zum laufen zu bekommen.

Jetzt hänge ich an dem Punkt die erste "test" extension für das interne isdn zu erstellen.

Jedoch bekomme ich keine Antwort vom Asterisk, obwohl er den Anruf registriert (log s.u.)
Irgendwie scheint er die extension.conf nicht lesen zu wollen ...

Die Dateien liegen (wie standart) in /etc/asterisk

Das ausgabelog:
Code:

  == Parsing '/etc/asterisk/asterisk.conf': Found

  == Parsing '/etc/asterisk/extconfig.conf': Found

Asterisk SVN-trunk-r10432, Copyright (C) 1999 - 2006 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.

=========================================================================

  == Parsing '/etc/asterisk/logger.conf': Found

Asterisk Event Logger Started /var/log/asterisk/event_log

Asterisk Dynamic Loader loading preload modules:

  == Parsing '/etc/asterisk/modules.conf': Found

  == Manager registered action Ping

  == Manager registered action Events

  == Manager registered action Logoff

  == Manager registered action Hangup

  == Manager registered action Status

  == Manager registered action Setvar

  == Manager registered action Getvar

  == Manager registered action Redirect

  == Manager registered action Originate

  == Manager registered action Command

  == Manager registered action ExtensionState

  == Manager registered action AbsoluteTimeout

  == Manager registered action MailboxStatus

  == Manager registered action MailboxCount

  == Manager registered action ListCommands

  == Parsing '/etc/asterisk/manager.conf': Found
Feb 19 16:51:54 NOTICE[10463]: cdr.c:1157 do_reload: CDR simple logging enabled.

  == Parsing '/etc/asterisk/rtp.conf': Found

  == RTP Allocating from port range 10000 -> 20000

  == UDPTL allocating from port range 4500 -> 4999

Asterisk PBX Core Initializing

Registering builtin applications:

 nswer]

  == Registered application 'Answer'

 ackGround]

  == Registered application 'BackGround'

 usy]

  == Registered application 'Busy'

 ongestion]

  == Registered application 'Congestion'

 oto]

  == Registered application 'Goto'

 otoIf]

  == Registered application 'GotoIf'

 otoIfTime]

  == Registered application 'GotoIfTime'

 xecIfTime]

  == Registered application 'ExecIfTime'

 angup]

  == Registered application 'Hangup'

 oOp]

  == Registered application 'NoOp'

 rogress]

  == Registered application 'Progress'

 esetCDR]

  == Registered application 'ResetCDR'

 inging]

  == Registered application 'Ringing'

 ayNumber]

  == Registered application 'SayNumber'

 ayDigits]

  == Registered application 'SayDigits'

 ayAlpha]

  == Registered application 'SayAlpha'

 ayPhonetic]

  == Registered application 'SayPhonetic'

 etAMAFlags]

  == Registered application 'SetAMAFlags'

 etGlobalVar]

  == Registered application 'SetGlobalVar'

 et]

  == Registered application 'Set'

 mportVar]

  == Registered application 'ImportVar'

 ait]

  == Registered application 'Wait'

 aitExten]

  == Registered application 'WaitExten'

Asterisk Dynamic Loader Starting:

  == Parsing '/etc/asterisk/modules.conf': Found

 [chan_capi.so] => (Common ISDN API for Asterisk)

  == Parsing '/etc/asterisk/capi.conf': Found

    -- capi_pvt ISDN1-pseudo-D (29,incoming,0,2) (0,4,64)

    -- capi_pvt ISDN1 (29,incoming,0,2) (0,4,64)

    -- capi_pvt ISDN1 (29,incoming,0,2) (0,4,64)

    -- listening on contr1 CIPmask = 0x1fff03ff

  == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.4) )

  == Registered application 'capiCommand'

  == Registered custom function VANITYNUMBER

 [res_musiconhold.so] => (Music On Hold Resource)

  == Registered application 'MusicOnHold'

  == Registered application 'WaitMusicOnHold'

  == Registered application 'SetMusicOnHold'

  == Registered application 'StartMusicOnHold'

  == Registered application 'StopMusicOnHold'

  == Parsing '/etc/asterisk/musiconhold.conf': Found
Feb 19 16:51:54 WARNING[10463]: res_musiconhold.c:988 load_moh_classes: The old musiconhold.conf syntax has been deprecated!  Please refer to the sample configuration for information on the new syntax.
Feb 19 16:51:54 WARNING[10463]: res_musiconhold.c:831 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.

  == Manager registered action DBGet

  == Manager registered action DBPut

  == Parsing '/etc/asterisk/enum.conf': Found

Asterisk Ready.
]1;Asterisk]2;Asterisk Console on 'telanlage' (pid 10463)*CLI> Feb 19 16:52:01 NOTICE[10467]: pbx.c:1597 pbx_extension_helper: Cannot find extension context 'incoming'
Feb 19 16:52:01 NOTICE[10467]: chan_capi.c:2218 start_pbx_on_match: ISDN1: did not find exten for '29', ignoring call.
*CLI> show dialplan
-= 0 extensions (0 priorities) in 0 contexts. =-
*CLI> show config mappings




*CLI> stop now
Beginning asterisk shutdown....

Executing last minute cleanups

  == Destroying musiconhold processes

Asterisk cleanly ending (0).

Wie man erkennt, kommt bei der abfrage des Dailplan nichts raus ...

Hier noch kurz die capi.conf
Code:
[general]
nationalprefix=0
internationalprefix=00
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=msn     ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
                 ;when using NT-mode, 'DID' should be set in any case
incomingmsn=29    ;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=fromisdn    ;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)

und die "irgendwie fehlende" extension.conf
Code:
[general]
static=yes
writeprotect=no

[default]
exten => 29,1,Wait,1
exten => 29,2,Answer			; Answer the line
exten => 29,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => 29,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => 29,5,BackGround(demo-congrats)	; Play a congratulatory message
exten => 29,6,BackGround(demo-instruct)	; Play some instructions

[fromisdn]
exten => s,1,Wait,1			; Wait a second, just for fun
exten => s,2,Answer			; Answer the line
exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)	; Play some instructions
include => default

Hat wer eine Idee woran es liegen könnte ...
 
Warum muss es den die aktuelle SVN Version sein?
Die kann auch einen Fehler enthalten.


Vielleicht solltest du erstmal Asterisk 1.2.4 testen.


Anonsten:
Schreibfehler? Zugriffsrechte?
Was passiert bei "extensions reload" ?
 
Die Datei heißt extensions.conf, nicht extension.conf, ist das evtl. der Fehler?

Gruß,
 
eine s-extension für isdn funktioniert meines wissens nicht. die msn muss als exten angegeben werden.
 
Die Datei heisst auch extensions.conf, war ein tippfehler ...

Habs in der zwischenzeit hinbekommen ...

Entweder wars ein Fehler in den Configs oder ein Bug in der svn version, denn nach dem neuinstallieren mit 1.2.4 wurde mir auch geantwortet ...

Ich hatte die SVN installiert, da ich zu der eine einigermassen vernünftige Beschreibung gefunden hatte, wie ich Asterisk mit Capi installiere ...

Ob die s-extension fuktioniert weis ich nicht, aber mit dem Include-Befehl tut es auf jeden fall

Danke für die Hilfe !!
 
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