[Problem] Asterisk nimmt eingehende ISDN Anrufe nicht an

linuzer

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Hallo Experten-Gemeinde,

mein Problem ist, dass eingehende ISDN Anrufe zwar scheinbar bei Asterisk ankommen, aber nicht weiter verarbeitet werden.

Ich habe bereits versucht, die Konfiguration so weit es geht zu vereinfachen:

capi.conf:
Code:
[general]
nationalprefix = 0
internationalprefix = 00
rxgain = 1.0
txgain = 1.0
language = de

[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
;accountcode=
context=isdn-in
immediate=yes
;echocancel=yes
;echocancelold=yes
callgroup=1
devices=2

sip.conf
Code:
[general]
bindport=5061
bindaddr=0.0.0.0
externhost=irgendwas.dynalias.org
externrefresh=10
nat=yes
canreinvite=no
srvlookup=yes
dtmfmode=info
language=de
disallow=all
allow=ulaw
allow=alaw

[30] 
callerid="30"
secret=1234 
type=friend 
host=dynamic
domain=192.168.0.1

extensions.conf
Code:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[default]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()

; Telefonie intern
exten => _X.,1,NoCDR()
exten => _X.,2,Dial(SIP/${EXTEN},20)
exten => _X.,3,VoiceMail(90,u)

; Telefonie nach extern
exten => _0X.,1,Dial(CAPI/ISDN1/${EXTEN})
exten => _0X.,2,Hangup

[isdn-in]
exten => _X.,1,Answer()
exten => _X.,2,Playback(hello-world)
exten => _X.,3,Hangup()

;exten => s,1,Answer()
;exten => s,2,Playback(hello-world)
;exten => s,3,Hangup()

;exten => _X.,1,Dial(SIP/30,20)
;exten => _X.,2,VoiceMail(90,u)

Wenn ich jetzt von extern anrufe, sieht das "capi debug" (bei höchstem Verbose-Level) so aus:
Code:
 CAPI: ApplId=0x0800 Command=0x02 SubCommand=0x82 MsgNum=0xf301 NCCI=0x01090000
CONNECT_IND                ID=2048 #0xf301 LEN=12288
  Controller/PLCI/NCCI            = 0x1090000
  CIPValue                        = 0x1000
  CalledPartyNumber               = <c1>[COLOR="blue"]1234567 [I](hier steht meine korrekte MSN)[/I][/COLOR]
  CallingPartyNumber              = <21 83>[COLOR="blue"]3234543 [I](hier steht meine korrekte Handy-Nr von der ich anrufe)[/I][/COLOR]
  CalledPartySubaddress           = default
  CallingPartySubaddress          = default
  BC                              = <80 90 a3>
  LLC                             = default
  HLC                             = <91 81>
  AdditionalInfo                  = default

    -- CONNECT_IND (PLCI=0,DID=[COLOR="blue"]1234567[/COLOR],CID=[COLOR="blue"]3234543[/COLOR],CIP=0x1000,CONTROLLER=0)
[Jan  1 00:56:10] [B]WARNING[5204]: chan_capi.c:4871 capidev_handle_connect_indication: did not find device for msn = [COLOR="blue"]1234567 [/COLOR][/B]
CONNECT_RESP               ID=2048 #0x01f3 LEN=4864
  Controller/PLCI/NCCI            = 0x901
  Reject                          = 0x100
  BProtocol                       = default
  ConnectedNumber                 = default
  ConnectedSubaddress             = default
  LLC                             = default
  AdditionalInfo                  = default

       > CAPI: Command=CONNECT_IND,0x8482: no interface for PLCI=0, MSGNUM=0xf301!
CAPI: ApplId=0x0800 Command=0x08 SubCommand=0x82 MsgNum=0xf401 NCCI=0x01090000
INFO_IND                   ID=2048 #0xf401 LEN=5888
  Controller/PLCI/NCCI            = 0x1090000
  InfoNumber                      = 0x7000
  InfoElement                     = <c1>[COLOR="blue"]1234567 [/COLOR]

INFO_RESP                  ID=2048 #0x01f4 LEN=3072
  Controller/PLCI/NCCI            = 0x0

CAPI: INFO_IND no interface for PLCI=0
       > CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0, MSGNUM=0xf401!
CAPI: ApplId=0x0800 Command=0x08 SubCommand=0x82 MsgNum=0xf501 NCCI=0x01090000
INFO_IND                   ID=2048 #0xf501 LEN=4096
  Controller/PLCI/NCCI            = 0x1090000
  InfoNumber                      = 0x1800
  InfoElement                     = <89>

INFO_RESP                  ID=2048 #0x01f5 LEN=3072
  Controller/PLCI/NCCI            = 0x0

CAPI: INFO_IND no interface for PLCI=0
       > CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0, MSGNUM=0xf501!
CAPI: ApplId=0x0800 Command=0x08 SubCommand=0x82 MsgNum=0xf601 NCCI=0x01090000
INFO_IND                   ID=2048 #0xf601 LEN=4096
  Controller/PLCI/NCCI            = 0x1090000
  InfoNumber                      = 0xa100
  InfoElement                     = <a1>

INFO_RESP                  ID=2048 #0x01f6 LEN=3072
  Controller/PLCI/NCCI            = 0x0

CAPI: INFO_IND no interface for PLCI=0
       > CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0, MSGNUM=0xf601!
CAPI: ApplId=0x0800 Command=0x08 SubCommand=0x82 MsgNum=0xf701 NCCI=0x01090000
INFO_IND                   ID=2048 #0xf701 LEN=3840
  Controller/PLCI/NCCI            = 0x1090000
  InfoNumber                      = 0xc0
  InfoElement                     = default

INFO_RESP                  ID=2048 #0x01f7 LEN=3072
  Controller/PLCI/NCCI            = 0x0

CAPI: INFO_IND no interface for PLCI=0
       > CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0, MSGNUM=0xf701!
CAPI: ApplId=0x0800 Command=0x04 SubCommand=0x82 MsgNum=0x0a02 NCCI=0x01090000
DISCONNECT_IND             ID=2048 #0x0a02 LEN=3584
  Controller/PLCI/NCCI            = 0x1090000
  Reason                          = 0x0

DISCONNECT_RESP            ID=2048 #0x020a LEN=3072
  Controller/PLCI/NCCI            = 0x0

CAPI: DISCONNECT_IND no interface for PLCI=0
       > CAPI: Command=DISCONNECT_IND,0x848c: no interface for PLCI=0, MSGNUM=0xa02!

Also wie es scheint, bekommt Asterisk zumindest mit, dass ein Anruf eingeht. Es gibt aber keinen Hinweis, der auf irgendeinen Dialplan oder so hindeuted, fast so, als ob er die extensions.conf nicht findet.

Natürlich habe ich auch die Warnung gesehen "did not find device for msn" -- allerdings konnte ich selbst nach langem Googlen keine weiterführenden Hinweise erhalten, was genau da schief geht (falls das überhaupt von Bedeutung ist).

Ich sollte noch erwähnen, dass ausgehende Anrufe von SIP/30 ans Handy möglich sind, wenngleich der Asterisk auch nicht auflegt, wenn SIP/30 aufgelegt hat -- aber das ist ein anderes Problem. Ausserdem kann ich interne Telefonate (SIP/30 und SIP/20) führen, wenn das SIP/20 in der sip.conf steht.

Hat jemand eine Idee, warum der Anruf von extern schief geht?
Vielen Dank!
 
Hast Du schon ausprobiert, die exten=>s zu verwenden? Durch das immediate=yes wird Asterisk vermutlich dort hin springen. Genaueres sollte ein CLI Auszug mit verbose>=3 (asterisk -rvvv) zeigen.
 
Hast Du schon ausprobiert, die exten=>s zu verwenden? Genaueres sollte ein CLI Auszug mit verbose>=3 (asterisk -rvvv) zeigen.

Ja, habe ich. Das sind die auskommentierten Teile in der extensions.conf. Das sind verschiedene Varianten, die ich alle ausprobiert habe.
Der letzte Auszug ist direkt von der CLI mit höchster Verbose und Debug Level. Wenn ich "capi debug" nicht aktiviere, passiert da überhaupt nichts - nichtmal 'ne Debug-Meldung, gar nichts. Nur wenn "capi debug" eingeschaltet ist, kommen obige Meldungen.
Der Fehler muss also schon an einer sehr frühen Stelle der Anrufverarbeitung passieren, irgendwo in den Tiefen von CAPI.
 
bringt Dich das weiter?

Code:
[isdn-in]
exten => _1234567,1,Answer()
exten => _1234567,2,Playback(hello-world)
exten => _1234567,3,Hangup()
 
Nein, das klappt leider auch nicht :-(
 
ich schicke Dir mal meine settings.
wenn du das so kopierst, würde ich sagen muss das gehen :)


Code:
[general]
nationalprefix=0
internationalprefix=00
;rxgain=1.0 ;original      ;linear receive gain (1.0 = no change)
;txgain=1.0 ;original      ;linear transmit gain (1.0 = no change)
rxgain=0.8 ;test hall
txgain=0.8 ;test hall
allow=all

[ISDN1]          ; fritzbox 7050/7170 external S0 (or external analog line: experimental)
ntmode=no      ;if isdn card operates in nt mode, set this to yes
ISDNmode=MSN    ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs,
                       ;a ="analog controller": empty incoming msn gets replaced
                       ;with defaultcid (-> fritzbox 7050/7170 at analog line)
;defaultcid=  ;set a default caller id to that interface for dial-out,
                 ;this caller id will be used when dial option 'd' is set.
controller=1     ;capi controller number to use (=4: fritzbox 7050/7150 at analog line)
group=1          ;dialout group
callgroup=1
pickupgroup=1
softdtmf=off      ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=off     ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode=     ;PBX accountcode to use in CDRs
context=isdn-in  ;context for incoming calls
;holdtype=hold   ;when the PBX puts the call on hold, ISDN HOLD will be used. If
                 ;set to 'local' (default value), no hold is done and the PBX may
                 ;play MOH.
bridge=no      ;native bridging (CAPI line interconnect) if available
devices=2        ;number of concurrent calls on this controller
                 ;(2 makes sense for single BRI, 30 for PRI)

Code:
[isdn-in]
exten => _X.,1,Answer()
exten => _X.,2,Playback(hello-world)
exten => _X.,3,Hangup()

Beachte dass Modifizierungen der Konfiguration von "chan_capi" nicht durch "module reload" aktuallisiert werden kann. Stoppen und starten von Asterisk ist dafür nötig.
 
Leider nicht, gleicher Fehler.

Aber lass mich mal was anderes fragen: Deiner Signatur entnehme ich, dass das bei Dir auf einer 7270 läuft, richtig? Ich vermute nämlich schon seit einiger Zeit, dass der Fehler mit der 7390 und ihrer (vielleicht anderen?) ISDN-Controller zu tun hat.

Deswegen mal die Frage an alle: Hat irgendwer schonmal chan_capi auf der 7390 zum laufen gekriegt? Wenn ja, würden mich da gerne mal die Details interessieren, wie chan_capi Version, capi.conf, Asterisk Version, und vor allem das Prozedere mit dem Asterisk kompiliert/installiert wurde.

LG linuzer
 
Wie sieht denn der CAPI betreffende Teil aus, wenn Du asterisk als Console startest? Vielleicht sieht man da etwas über die vorhandenen Channels.
 
Hier ist der komplette Startvorgang:

Code:
root@fritz:/var/media/ftp/Storage-01/external# asterisk -vvvvvvvdddddddc /etc/as
terisk/asterisk.conf
Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf
  == Found
Seeding global EID '00:24:fe:ed:11:4f' from 'eth0' using 'siocgifhwaddr'
  == Parsing '/etc/asterisk/logger.conf': Parsing /etc/asterisk/logger.conf
  == Found
 Asterisk Event Logger Started /var/lib/asterisk/log/event_log
 Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf':   == Found
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Login
  == Manager registered action Challenge
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action GetConfig
  == Manager registered action GetConfigJSON
  == Manager registered action UpdateConfig
  == Manager registered action CreateConfig
  == Manager registered action ListCategories
  == Manager registered action Redirect
  == Manager registered action Atxfer
  == 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
  == Manager registered action SendText
  == Manager registered action UserEvent
  == Manager registered action WaitEvent
  == Manager registered action CoreSettings
  == Manager registered action CoreStatus
  == Manager registered action Reload
  == Manager registered action CoreShowChannels
  == Manager registered action ModuleLoad
  == Manager registered action ModuleCheck
  == Parsing '/etc/asterisk/manager.conf':   == Found
  == Parsing '/etc/asterisk/cdr.conf':   == Found
[Sep 28 13:09:56] NOTICE[8477]: cdr.c:1484 do_reload: CDR simple logging enabled.
  == Parsing '/etc/asterisk/rtp.conf':   == Found
  == RTP Allocating from port range 7111 -> 7142
       > Can't find dsp config file dsp.conf. Assuming default silencethreshold of 256.
 Asterisk PBX Core Initializing
 Registering builtin applications:
  == Registered custom function 'EXCEPTION'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [ExecIfTime]
  == Registered application 'ExecIfTime'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [ImportVar]
  == Registered application 'ImportVar'
 [Hangup]
  == Registered application 'Hangup'
 [Incomplete]
  == Registered application 'Incomplete'
 [NoOp]
  == Registered application 'NoOp'
 [Proceeding]
  == Registered application 'Proceeding'
 [Progress]
  == Registered application 'Progress'
 [RaiseException]
  == Registered application 'RaiseException'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [Ringing]
  == Registered application 'Ringing'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [Set]
  == Registered application 'Set'
 [MSet]
  == Registered application 'MSet'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
  == Manager registered action ShowDialPlan
  == Parsing '/etc/asterisk/indications.conf':   == Found
    -- Registered indication country 'de'
    -- Setting default indication country to 'de'
  == Registered application 'Bridge'
    -- Registered extension context 'parkedcalls' (0x575ff0) in table 0x575fa0; registrar: features
  == Parsing '/etc/asterisk/features.conf':   == Found
    -- Registered extension context 'parkedcalls' (0x576078) in table 0x575fa0; registrar: features
    -- Added extension '700' priority 1 to parkedcalls (0x576078)
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
  == Manager registered action Park
  == Manager registered action Bridge
  == Manager registered action DBGet
  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
 Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf':   == Found
[Sep 28 13:09:56] NOTICE[8477]: loader.c:1060 load_modules: 76 modules will be loaded.
  == AGI Command 'answer' registered
  == AGI Command 'asyncagi break' registered
  == AGI Command 'channel status' registered
  == AGI Command 'database del' registered
  == AGI Command 'database deltree' registered
  == AGI Command 'database get' registered
  == AGI Command 'database put' registered
  == AGI Command 'exec' registered
  == AGI Command 'get data' registered
  == AGI Command 'get full variable' registered
  == AGI Command 'get option' registered
  == AGI Command 'get variable' registered
  == AGI Command 'hangup' registered
  == AGI Command 'noop' registered
  == AGI Command 'receive char' registered
  == AGI Command 'receive text' registered
  == AGI Command 'record file' registered
  == AGI Command 'say alpha' registered
  == AGI Command 'say digits' registered
  == AGI Command 'say number' registered
  == AGI Command 'say phonetic' registered
  == AGI Command 'say date' registered
  == AGI Command 'say time' registered
  == AGI Command 'say datetime' registered
  == AGI Command 'send image' registered
  == AGI Command 'send text' registered
  == AGI Command 'set autohangup' registered
  == AGI Command 'set callerid' registered
  == AGI Command 'set context' registered
  == AGI Command 'set extension' registered
  == AGI Command 'set music' registered
  == AGI Command 'set priority' registered
  == AGI Command 'set variable' registered
  == AGI Command 'stream file' registered
  == AGI Command 'control stream file' registered
  == AGI Command 'tdd mode' registered
  == AGI Command 'verbose' registered
  == AGI Command 'wait for digit' registered
  == AGI Command 'speech create' registered
  == AGI Command 'speech set' registered
  == AGI Command 'speech destroy' registered
  == AGI Command 'speech load grammar' registered
  == AGI Command 'speech unload grammar' registered
  == AGI Command 'speech activate grammar' registered
  == AGI Command 'speech deactivate grammar' registered
  == AGI Command 'speech recognize' registered
  == Registered application 'DeadAGI'
  == Registered application 'EAGI'
  == Manager registered action AGI
  == Registered application 'AGI'
 res_agi.so => (Asterisk Gateway Interface (AGI))
  == Registered file format g726-40, extension(s) g726-40
  == Registered file format g726-32, extension(s) g726-32
  == Registered file format g726-24, extension(s) g726-24
  == Registered file format g726-16, extension(s) g726-16
 format_g726.so => (Raw G.726 (16/24/32/40kbps) data)
  == Registered file format g729, extension(s) g729
 format_g729.so => (Raw G729 data)
  == Registered file format gsm, extension(s) gsm
 format_gsm.so => (Raw GSM data)
  == Registered file format h263, extension(s) h263
 format_h263.so => (Raw H.263 data)
  == Registered file format h264, extension(s) h264
 format_h264.so => (Raw H.264 data)
  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu|ulw
  == Registered file format alaw, extension(s) alaw|al|alw
  == Registered file format au, extension(s) au
  == Registered file format g722, extension(s) g722
 format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)
  == Registered file format sln, extension(s) sln|raw
 format_sln.so => (Raw Signed Linear Audio support (SLN))
  == Registered file format sln16, extension(s) sln16
 format_sln16.so => (Raw Signed Linear 16KHz Audio support (SLN16))
  == Registered file format wav, extension(s) wav
 format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))
  == Registered file format wav49, extension(s) WAV|wav49
 format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM))
  == Parsing '/etc/asterisk/iax.conf':   == Found
  == Binding IAX2 to default address 0.0.0.0:4569
  == Registered application 'IAX2Provision'
  == Registered custom function 'IAXPEER'
  == Registered custom function 'IAXVAR'
  == Manager registered action IAXpeers
  == Manager registered action IAXpeerlist
  == Manager registered action IAXnetstats
  == Manager registered action IAXregistry
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == 10 helper threads started
  == IAX Ready and Listening
[Sep 28 13:09:59] NOTICE[8477]: iax2-provision.c:553 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.
 chan_iax2.so => (Inter Asterisk eXchange (Ver 2))
  == Parsing '/etc/asterisk/capi.conf':   == Found
    -- Registering at CAPI (blocksize=160 maxlogicalchannels=2)
  == This box has 5 capi controller(s).
    -- Contr1 supports DTMF
    -- Contr1 supports supplementary services
       > FACILITY_CONF INFO = 0
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300
    -- Contr1 private options=0x01000001
    -- Contr2 supports DTMF
    -- Contr2 supports supplementary services
       > FACILITY_CONF INFO = 0
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300
    -- Contr2 private options=0x01000001
    -- Contr3 supports DTMF
    -- Contr3 supports supplementary services
       > FACILITY_CONF INFO = 0
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300
    -- Contr3 private options=0x00000001
    -- Contr4 supports DTMF
    -- Contr4 supports supplementary services
       > FACILITY_CONF INFO = 0
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300
    -- Contr4 private options=0x00000001
    -- Contr5 supports DTMF
    -- Contr5 supports supplementary services
       > FACILITY_CONF INFO = 0
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300
    -- Contr5 private options=0x00000001
  == Reading config for ISDN1
    -- capi D ISDN1#00 (*:isdn-in) contr=1 devs=2 EC=0,opt=4,tail=0
    -- capi B ISDN1#01 (*:isdn-in) contr=1 devs=2 EC=0,opt=4,tail=0
    -- capi B ISDN1#02 (*:isdn-in) contr=1 devs=2 EC=0,opt=4,tail=0
    -- Registering at CAPI (blocksize=160 maxlogicalchannels=3)
    -- listening on contr1 CIPmask = 0x1fff03ff
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7999 cc_post_init_capi: Unused contr2
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7999 cc_post_init_capi: Unused contr3
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7999 cc_post_init_capi: Unused contr4
[Sep 28 13:09:59] NOTICE[8477]: chan_capi.c:7999 cc_post_init_capi: Unused contr5
  == Registered channel type 'CAPI' (Common ISDN API Driver (1.1.5))
  == Registered application 'capicommand'
[Sep 28 13:09:59] NOTICE[8497]: chan_capi.c:7069 capidev_loop: Started CAPI device thread for CAPI Appl-ID 7.
 chan_capi.so => (Common ISDN API Driver (1.1.5))
  == Parsing '/etc/asterisk/cdr_sqlite3_custom.conf':   == Found
    -- cdr_sqlite3_custom: Logging CDR records to table 'cdr' in 'master.db'
 cdr_sqlite3_custom.so => (SQLite3 Custom CDR Module)
  == Parsing '/etc/asterisk/cdr_manager.conf':   == Found
 cdr_manager.so => (Asterisk Manager Interface CDR Backend)
  == Parsing '/etc/asterisk/cdr.conf':   == Found
 cdr_csv.so => (Comma Separated Values CDR Backend)
  == Registered application 'WaitForRing'
 app_waitforring.so => (Waits until first ring after time)
  == Parsing '/etc/asterisk/voicemail.conf':   == Found
  == Registered application 'VoiceMail'
  == Registered application 'VoiceMailMain'
  == Registered application 'MailboxExists'
  == Registered application 'VMAuthenticate'
  == Registered custom function 'MAILBOX_EXISTS'
  == Manager registered action VoicemailUsersList
 app_voicemail.so => (Comedian Mail (Voicemail System))
  == Registered application 'Transfer'
 app_transfer.so => (Transfers a caller to another extension)
  == Registered application 'TrySystem'
  == Registered application 'System'
 app_system.so => (Generic System() application)
  == Registered application 'SoftHangup'
 app_softhangup.so => (Hangs up the requested channel)
  == Parsing '/etc/asterisk/musiconhold.conf':   == Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
  == Registered application 'StartMusicOnHold'
  == Registered application 'StopMusicOnHold'
 res_musiconhold.so => (Music On Hold Resource)
 res_curl.so => (cURL Resource Module)
 res_adsi.so => (ADSI Resource)
 pbx_spool.so => (Outgoing Spool Support)
 pbx_realtime.so => (Realtime Switch)
  == Parsing '/etc/asterisk/extensions.conf':   == Found
    -- Registered extension context 'default' (0x5e1398) in local table 0x5e1690; registrar: pbx_config
    -- Added extension '1001' priority 1 to default (0x5e1398)
    -- Added extension '1001' priority 2 to default (0x5e1398)
    -- Added extension '1001' priority 3 to default (0x5e1398)
    -- Added extension '_X.' priority 1 to default (0x5e1398)
    -- Added extension '_X.' priority 2 to default (0x5e1398)
    -- Added extension '_X.' priority 3 to default (0x5e1398)
    -- Added extension '_0X.' priority 1 to default (0x5e1398)
    -- Added extension '_0X.' priority 2 to default (0x5e1398)
    -- Registered extension context 'VonDraussen' (0x617610) in local table 0x5e1690; registrar: pbx_config
    -- Added extension '620' priority 1 to VonDraussen (0x617610)
    -- Added extension '620' priority 2 to VonDraussen (0x617610)
    -- Registered extension context 'isdn-in' (0x617fb8) in local table 0x5e1690; registrar: pbx_config
    -- Registered extension context 'parkedcalls' (0x5e05f8) in local table 0x5e1690; registrar: features
    -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config
    -- Added extension '700' priority 1 to parkedcalls (0x5e05f8)
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000000 sec
    -- Time to restore hints and swap in new dialplan: 0.000000 sec
    -- Time to delete the old dialplan: 0.000000 sec
    -- Total time merge_contexts_delete: 0.000000 sec
 pbx_config.so => (Text Extension Configuration)
  == Registered custom function 'VOLUME'
 func_volume.so => (Technology independent volume control)
  == Registered custom function 'TIMEOUT'
 func_timeout.so => (Channel timeout dialplan functions)
  == Registered custom function 'FIELDQTY'
  == Registered custom function 'FILTER'
  == Registered custom function 'LISTFILTER'
  == Registered custom function 'REGEX'
  == Registered custom function 'ARRAY'
  == Registered custom function 'QUOTE'
  == Registered custom function 'CSV_QUOTE'
  == Registered custom function 'LEN'
  == Registered custom function 'STRFTIME'
  == Registered custom function 'STRPTIME'
  == Registered custom function 'EVAL'
  == Registered custom function 'KEYPADHASH'
  == Registered custom function 'HASHKEYS'
  == Registered custom function 'HASH'
  == Registered application 'ClearHash'
  == Registered custom function 'TOUPPER'
  == Registered custom function 'TOLOWER'
 func_strings.so => (String handling dialplan functions)
  == Registered custom function 'IFMODULE'
 func_module.so => (Checks if Asterisk module is loaded in memory)
  == Registered custom function 'MD5'
 func_md5.so => (MD5 digest dialplan functions)
  == Registered custom function 'MATH'
 func_math.so => (Mathematical dialplan function)
  == Registered custom function 'ISNULL'
  == Registered custom function 'SET'
  == Registered custom function 'EXISTS'
  == Registered custom function 'IF'
  == Registered custom function 'IFTIME'
  == Registered custom function 'IMPORT'
 func_logic.so => (Logical dialplan functions)
  == Registered custom function 'GROUP_COUNT'
  == Registered custom function 'GROUP_MATCH_COUNT'
  == Registered custom function 'GROUP_LIST'
  == Registered custom function 'GROUP'
 func_groupcount.so => (Channel group dialplan functions)
  == Registered custom function 'GLOBAL'
  == Registered custom function 'SHARED'
 func_global.so => (Variable dialplan functions)
  == Registered custom function 'ENV'
  == Registered custom function 'STAT'
  == Registered custom function 'FILE'
 func_env.so => (Environment/filesystem dialplan functions)
  == Registered custom function 'ENUMRESULT'
  == Registered custom function 'ENUMQUERY'
  == Registered custom function 'ENUMLOOKUP'
  == Registered custom function 'TXTCIDNAME'
 func_enum.so => (ENUM related dialplan functions)
  == Registered custom function 'DIALGROUP'
 func_dialgroup.so => (Dialgroup dialplan function)
  == Registered custom function 'DB'
  == Registered custom function 'DB_EXISTS'
  == Registered custom function 'DB_DELETE'
 func_db.so => (Database (astdb) related dialplan functions)
  == Registered custom function 'CUT'
  == Registered custom function 'SORT'
 func_cut.so => (Cut out information from a string)
  == Registered custom function 'CURL'
  == Registered custom function 'CURLOPT'
 func_curl.so => (Load external URL)
  == Registered custom function 'AST_CONFIG'
 func_config.so => (Asterisk configuration file variable access)
  == Registered custom function 'CHANNEL'
  == Registered custom function 'CHANNELS'
 func_channel.so => (Channel information dialplan functions)
  == Registered custom function 'CDR'
 func_cdr.so => (Call Detail Record (CDR) dialplan function)
  == Registered custom function 'CALLERPRES'
  == Registered custom function 'CALLERID'
 func_callerid.so => (Caller ID related dialplan functions)
  == Registered translator 'alawtolin' from format alaw to slin, cost 1
  == Registered translator 'lintoalaw' from format slin to alaw, cost 1
 codec_alaw.so => (A-law Coder/Decoder)
  == Registered application 'MixMonitor'
  == Registered application 'StopMixMonitor'
 app_mixmonitor.so => (Mixed Audio Monitoring Application)
  == Registered application 'ConfBridge'
 app_confbridge.so => (Conference Bridge Application)
  == Registered application 'MacroExit'
  == Registered application 'MacroIf'
  == Registered application 'MacroExclusive'
  == Registered application 'Macro'
 app_macro.so => (Extension Macros)
  == Registered translator 'gsmtolin' from format gsm to slin, cost 8000
  == Registered translator 'lintogsm' from format slin to gsm, cost 12001
 codec_gsm.so => (GSM Coder/Decoder)
  == Registered application 'ForkCDR'
 app_forkcdr.so => (Fork The CDR into 2 separate entities)
  == Registered application 'ChanSpy'
  == Registered application 'ExtenSpy'
 app_chanspy.so => (Listen to the audio of an active channel)
  == Registered application 'Exec'
  == Registered application 'TryExec'
  == Registered application 'ExecIf'
 app_exec.so => (Executes dialplan applications)
  == Registered application 'Echo'
 app_echo.so => (Simple Echo Application)
  == Registered application 'SendText'
 app_sendtext.so => (Send Text Applications)
  == Manager registered action PlayDTMF
  == Registered application 'SendDTMF'
 app_senddtmf.so => (Send DTMF digits Application)
  == Registered application 'SayUnixTime'
  == Registered application 'DateTime'
 app_sayunixtime.so => (Say time)
  == Registered application 'Record'
 app_record.so => (Trivial Record Application)
  == Registered application 'PlayTones'
  == Registered application 'StopPlayTones'
 app_playtones.so => (Playtones Application)
  == Registered application 'Playback'
 app_playback.so => (Sound File Playback Application)
  == Registered application 'MP3Player'
 app_mp3.so => (Silly MP3 Application)
  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1
 codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder)
  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
 codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder)
  == Registered translator 'g726tolin' from format g726 to slin, cost 16001
  == Registered translator 'lintog726' from format slin to g726, cost 28002
  == Registered translator 'g726aal2tolin' from format g726aal2 to slin, cost 16001
  == Registered translator 'lintog726aal2' from format slin to g726aal2, cost 28001
 codec_g726.so => (ITU G.726-32kbps G726 Transcoder)
  == Registered application 'Authenticate'
 app_authenticate.so => (Authentication Application)
  == Registered application 'ChanIsAvail'
 app_chanisavail.so => (Check channel availability)
  == Registered application 'DISA'
 app_disa.so => (DISA (Direct Inward System Access) Application)
    -- Registered extension context 'app_dial_gosub_virtual_context' (0x616cb0) in table 0x5e1690; registrar: app_dial
    -- Added extension 's' priority 1 to app_dial_gosub_virtual_context (0x616cb0)
  == Registered application 'Dial'
  == Registered application 'RetryDial'
 app_dial.so => (Dialing Application)
  == Registered application 'DBdel'
  == Registered application 'DBdeltree'
 app_db.so => (Database Access Functions)
SIP channel loading...
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == SIP Listening on 0.0.0.0:5061
  == Using SIP CoS mark 4
[Sep 28 13:09:59] NOTICE[8477]: chan_sip.c:24785 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Registered application 'SIPAddHeader'
  == Registered application 'SIPRemoveHeader'
  == Registered custom function 'SIP_HEADER'
  == Registered custom function 'SIPPEER'
  == Registered custom function 'SIPCHANINFO'
  == Registered custom function 'CHECKSIPDOMAIN'
  == Manager registered action SIPpeers
  == Manager registered action SIPshowpeer
  == Manager registered action SIPqualifypeer
  == Manager registered action SIPshowregistry
  == Manager registered action SIPnotify
 chan_sip.so => (Session Initiation Protocol (SIP))
  == Registered translator 'g722tolin' from format g722 to slin, cost 12001
  == Registered translator 'lintog722' from format slin to g722, cost 8001
  == Registered translator 'g722tolin16' from format g722 to slin16, cost 16001
  == Registered translator 'lin16tog722' from format slin16 to g722, cost 16001
 codec_g722.so => (ITU G.722-64kbps G722 Transcoder)
  == Registered application 'SetCallerPres'
 app_setcallerid.so => (Set CallerID Presentation Application)
  == Registered application 'NoCDR'
 app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 chan_local.so => (Local Proxy Channel (Note: used internally by other modules))
  == Registered translator 'ulawtolin' from format ulaw to slin, cost 1
  == Registered translator 'lintoulaw' from format slin to ulaw, cost 1
 codec_ulaw.so => (mu-Law Coder/Decoder)
Asterisk Ready.
*CLI> [Sep 28 13:09:59] NOTICE[8500]: chan_sip.c:18912 handle_response_peerpoke: Peer 'NachDraussen' is now Reachable. (11ms / 2000ms)

Auffällig ist sind die Meldungen "chan_capi.c:7299 supported_sservices: unexpected FACILITY_SELECTOR = 0x300". Ich hab auch schon ausgiebig danach gegoogelt, aber leider ohne Erfolg. Auch im Sourcecode an der entsprechenden Zeile kann ich nicht wirklich erkennen, was wirklich schief geht.

Also inzwischen habe ich den Asterisk einfach über SIP mit der Fritzbox verbunden, was ganz gut zu funktionieren scheint. Es wundert mich halt nur, dass es nativ über chan_capi nicht geht, zumal es auf den anderen Boxen (7270, u.a.) offensichtilich funktioniert.

Mein Verdacht erhärtet sich immer mehr, dass das an der 7390 liegt -- ausser jemand kann das falsifizieren.

LG linuzer
 
Hm, der Startvorgang sieht imho brauchbar aus.

Ist bei der 7390 vielleicht der Controller 1 nicht für das Amt zuständig? :noidea: Ich finde bei Google auf die Schnelle nichts dazu.

Kannst Du mal probieren, die anderen Controller testweise auch zu konfigurieren? Ansonsten fällt mir nichts ein/auf.
 
Vielleicht eine Lösung

Hallo,

da ich das gleich Problem habe hier was gefunden.

Vielleicht kannst du damit ja was anfangen.

Es liegt wohl am an der libcapi die wohl nicht ersetzt werden darf.

Wenn mal einer Zeit und lusst hat.

Kann mir das mal einer erklären wie ich die Original LibCapi von der Fritz Firmware behalte.

gruß Pat2381
 
Hallo,

da ich das gleich Problem habe hier was gefunden.

Vielen Dank für den Hinweis, den hatte ich auch gefunden, bin aber genauso hängen geblieben. Vielleicht hab ich's nicht richtig gemacht, aber leider antwortet der ursprüngliche Poster nicht ... :(
Der könnte wohl am einfachsten eine detaillierte Anleitung geben...:habenwol:
 
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