Verschluckte Ziffern beim Callback!

Quetsch

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Hallo!

Ich nutze ein CallBack über Asterisk mit einer Fritz-ISDN Karte und ner Phoneflat. Nun habe ich von Suse 9.1 auf 10.0 aupgedatet, das alte asterisk-1.0.6-4 und asterisk-capi-0.3.5-5 installiert. (Aktuelle Version mit chan_capi 065 hab ich nicht zum Laufen gebracht... :-()
Das funzt soweit auch prima, Asterisk startet einwandfrei, der Callback wird ausgelöst. Wie gesagt, auf dem alten System lief alles einwandfrei!

ABER: starte ich die Ausgabe mit asterisk -vvvvvv sehe ich, daß gewählte Ziffern einfach verschluckt werden, ich teilweise 3-5 mal eine einzelne Ziffer eingeben muß bis dies erscheint:
sent CALLEDPARTYNUMBER INFO digit = 0 (PLCI=0x201)

Sind alle Ziffern durch, wird die Verbindung aufgebaut, nur werden Ziffern "verschluckt" und somit ist ein Callback von unterwegs unmöglich!

Wo liegt evtl. der Fehler?

Hier meine ganzen configs:

Code:
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=797302,797304
incomingmsn=797302
controller=1
softdtmf=1
accountcode=
context=capiin
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

extensions.conf:
Code:
[general]

static=yes
writeprotect=no


[default]
include => capiin

[capiin]
exten => 797302/0175xxxxxxx,1,Wait,1
exten => 797302/0175xxxxxxx,2,System(cp /etc/asterisk/callMichael /var/spool/asterisk/outgoing/)
exten => 797302/0175xxxxxxx,3,Hangup

[capidialtone]
exten => s,1,Dial,CAPI/@797302:b
exten => s,2,Hangup
include => capiin

[dialout]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,DISA(no-password|dial_now)
exten => s,4,Wait(2)
exten => s,5,Hangup
exten => s,102,Busy
exten => h,1,Hangup


[dial_now]
exten => _.,1,SetCallerID(07071xxxxxx)
exten => _.,2,Dial,CAPI/@797304:b
exten => _.,3,Congestion

Modules.conf:
Code:
;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; 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
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => chan_capi.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_modem.so=yes
chan_capi.so=yes

Und das CallFile CallMichael;
Code:
Channel: CAPI/797302:07071xxxxxx
Context: capidialtone
Extension: s
SetVar: CALLERIDNUM=797302
MaxRetries: 2
RetryTime: 10

PS: Tut mir leid, daß ich die Files falsch poste, hab auf die Schnelle nicht gefunden, wie sie richtig zitiert werden, editiere mein Post aber sofort auf Hinweis
 
Zuletzt bearbeitet:
Ich mache zwar kein Callback mit CAPI, würde aber mal testen ein
relaxdtmf=1
unter [interfaces] in der capi.conf hinzuzufügen. Wenn das nicht hilft, probiere softdtmf=0

Code setzt man in [ code ] .... [ /code ] tags (ohne die Leerzeichen).

Gruß,
Tin
 
Danke für die schnelle Antwort.

Leider bringt weder relaxdtmf=1 ohne softdtmf=0 und beide zusammen oder nur softdtmf=0 eine Verbesserung.
Es werden weiterhin einfach Ziffern "überhört"...

Woran kann das liegen, bitte helft!!

Hier mal meine Konsolenausgabe für asterisk -vvvvvv als root:
Code:
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect.
linux:/home/mkrueger # asterisk -vvvvvvv
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect.
linux:/home/mkrueger # /etc/init.d/asterisk stop
Shutting down Asterisk                                               done
linux:/home/mkrueger # asterisk -vvvvvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[email protected]>
=========================================================================
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == 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 DBget
  == Manager registered action DBput
  == Manager registered action DBdel
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 20000
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
 [chan_capi.so] => (Common ISDN API for Asterisk)
  == Parsing '/etc/asterisk/capi.conf': Found
    -- This box has 1 capi controller(s).
    -- CAPI[contr1] supports DTMF
    -- CAPI[contr1] supports supplementary services
       > sent FACILITY_REQ (CONTROLLER=0x1)
       > FACILITY_CONF INFO = 0
       > HOLD/RETRIEVE
       > TERMINAL PORTABILITY
       > ECT
       > 3PTY
       > CF
       > CD
       > MCID
       > CCBS
       > MWI
       > CCNR
  == ast_capi_pvt(797302,797304,797302,capiin,0,2) (1,2,64)
  == ast_capi_pvt(797302,797304,797302,capiin,0,2) (1,2,64)
    -- listening on contr1 CIPmask = 0x1fff03ff
  == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Jul 29 13:03:19 WARNING[10616]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [res_indications.so] => (Indications Configuration)
  == Parsing '/etc/asterisk/indications.conf': Found
    -- Registered indication country 'cl'
    -- Registered indication country 'tw'
    -- Registered indication country 'us'
    -- Registered indication country 'au'
    -- Registered indication country 'fr'
    -- Registered indication country 'de'
    -- Registered indication country 'nl'
    -- Registered indication country 'uk'
    -- Registered indication country 'fi'
    -- Registered indication country 'no'
    -- Registered indication country 'br'
    -- Registered indication country 'za'
    -- Registered indication country 'it'
    -- Registered indication country 'us-o'
    -- Registered indication country 'gr'
    -- Registered indication country 'ru'
    -- Registered indication country 'nz'
    -- Setting default indication country to 'us'
  == Registered application 'Playtones'
  == Registered application 'StopPlaytones'
 [res_features.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/features.conf': Found
    -- Registered extension context 'parkedcalls'
    -- Added extension '700' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
  == Registered application 'HoldedCall'
  == Registered application 'AutoanswerLogin'
  == Registered application 'Autoanswer'
 [res_agi.so] => (Asterisk Gateway Interface (AGI))
  == Registered application 'DeadAGI'
  == Registered application 'EAGI'
  == Registered application 'AGI'
 [res_crypto.so] => (Cryptographic Digital Signatures)
    -- Loaded PUBLIC key 'iaxtel'
    -- Loaded PUBLIC key 'freeworlddialup'
 [res_adsi.so] => (ADSI Resource)
  == Parsing '/etc/asterisk/adsi.conf': Found
 [res_monitor.so] => (Call Monitoring Resource)
  == Registered application 'Monitor'
  == Registered application 'StopMonitor'
  == Registered application 'ChangeMonitor'
  == Manager registered action Monitor
  == Manager registered action StopMonitor
  == Manager registered action ChangeMonitor
 [app_sms.so] => (SMS/PSTN handler)
  == Registered application 'SMS'
 [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.)
  == Registered application 'HasVoicemail'
  == Registered application 'HasNewVoicemail'
 [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
  == Registered file format wav49, extension(s) WAV|wav49
 [app_url.so] => (Send URL Applications)
  == Registered application 'SendURL'
 [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
 [app_test.so] => (Interface Test Application)
  == Registered application 'TestClient'
  == Registered application 'TestServer'
 [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2727
  == Using TOS bits 0
  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
 [app_eval.so] => (Reevaluates strings)
  == Registered application 'Eval'
 [app_sendtext.so] => (Send Text Applications)
  == Registered application 'SendText'
 [app_exec.so] => (Executes applications)
  == Registered application 'Exec'
 [app_txtcidname.so] => (TXTCIDName)
  == Registered application 'TXTCIDName'
  == Parsing '/etc/asterisk/enum.conf': Found
 [cdr_manager.so] => (Asterisk Call Manager CDR Backend)
  == Parsing '/etc/asterisk/cdr_manager.conf': Found
 [app_capiCD.so] => ((CAPI*) Call Deflection, the magic thing.)
  == Registered application 'capiCD'
 [app_directory.so] => (Extension Directory)
  == Registered application 'Directory'
 [app_playback.so] => (Trivial Playback Application)
  == Registered application 'Playback'
 [app_capiNoES.so] => ((CAPI*) No Echo Suppression.)
  == Registered application 'capiNoES'
 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
  == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1
  == Registered translator 'lintoadpcm' from format slin to adpcm, cost 4
 [chan_local.so] => (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [app_groupcount.so] => (Group Management Routines)
  == Registered application 'GetGroupCount'
  == Registered application 'SetGroup'
  == Registered application 'CheckGroup'
 [app_adsiprog.so] => (Asterisk ADSI Programming Application)
  == Registered application 'ADSIProg'
 [app_chanisavail.so] => (Check if channel is available)
  == Registered application 'ChanIsAvail'
 [app_qcall.so] => (Call from Queue)
 [app_softhangup.so] => (Hangs up the requested channel)
  == Registered application 'SoftHangup'
 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
  == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 9
  == Registered translator 'lintolpc10' from format slin to lpc10, cost 51
 [app_setcidname.so] => (Set CallerID Name)
  == Registered application 'SetCIDName'
 [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
  == 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_g729.so] => (Raw G729 data)
  == Registered file format g729, extension(s) g729
 [app_userevent.so] => (Custom User Event Application)
  == Registered application 'UserEvent'
 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
  == Registered translator 'gsmtolin' from format gsm to slin, cost 3
  == Registered translator 'lintogsm' from format slin to gsm, cost 8
 [app_authenticate.so] => (Authentication Application)
  == Registered application 'Authenticate'
 [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
  == Registered file format alaw, extension(s) alaw|al
 [format_ilbc.so] => (Raw iLBC data)
  == Registered file format iLBC, extension(s) ilbc
 [format_h263.so] => (Raw h263 data)
  == Registered file format h263, extension(s) h263
 [app_forkcdr.so] => (Fork The CDR into 2 separate entities.)
  == Registered application 'ForkCDR'
 [app_ices.so] => (Encode and Stream via icecast and ices)
  == Registered application 'ICES'
 [app_nbscat.so] => (Silly NBS Stream Application)
  == Registered application 'NBScat'
 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
  == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1
  == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1
 [app_system.so] => (Generic System() application)
  == Registered application 'TrySystem'
  == Registered application 'System'
 [app_record.so] => (Trivial Record Application)
  == Registered application 'Record'
 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
Jul 29 13:03:20 WARNING[10616]: chan_iax2.c:7487 load_module: Unable to open IAX timing interface: No such file or directory
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
 [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
  == Registered application 'Milliwatt'
 [app_parkandannounce.so] => (Call Parking and Announce Application)
  == Registered application 'ParkAndAnnounce'
 [app_sayunixtime.so] => (Say time)
  == Registered application 'SayUnixTime'
  == Registered application 'DateTime'
 [pbx_spool.so] => (Outgoing Spool Support)
 [app_capiMCID.so] => ((CAPI*) Malicious Caller ID, the evil thing.)
  == Registered application 'capiMCID'
 [app_macro.so] => (Extension Macros)
  == Registered application 'Macro'
 [app_random.so] => (Random goto)
  == Registered application 'Random'
 [codec_ulaw.so] => (Mu-law Coder/Decoder)
  == Registered translator 'ulawtolin' from format ulaw to slin, cost 1
  == Registered translator 'lintoulaw' from format slin to ulaw, cost 1
 [app_capiRETRIEVE.so] => ((CAPI*) RETRIEVE)
  == Registered application 'capiRETRIEVE'
 [chan_agent.so] => (Agent Proxy Channel)
  == Registered channel type 'Agent' (Call Agent Proxy Channel)
  == Registered application 'AgentLogin'
  == Registered application 'AgentCallbackLogin'
  == Registered application 'AgentMonitorOutgoing'
  == Parsing '/etc/asterisk/agents.conf': Found
 [app_controlplayback.so] => (Control Playback Application)
  == Registered application 'ControlPlayback'
 [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
  == Registered format 'jpg' (JPEG (Joint Picture Experts Group))
 [codec_alaw.so] => (A-law Coder/Decoder)
  == Registered translator 'alawtolin' from format alaw to slin, cost 1
  == Registered translator 'lintoalaw' from format slin to alaw, cost 1
 [app_transfer.so] => (Transfer)
  == Registered application 'Transfer'
 [cdr_csv.so] => (Comma Separated Values CDR Backend)
 [app_voicemail.so] => (Comedian Mail (Voicemail System))
  == Registered application 'VoiceMail'
  == Registered application 'VoiceMail2'
  == Registered application 'VoiceMailMain'
  == Registered application 'VoiceMailMain2'
  == Registered application 'MailboxExists'
  == Parsing '/etc/asterisk/voicemail.conf': Found
 [app_pickup.so] => (PickUp/PickDown/Steal/PickupChan)
  == Registered application 'PickupChan'
  == Registered application 'PickDown'
  == Registered application 'Steal'
  == Registered application 'PickUp'
 [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator)
  == Registered translator 'speextolin' from format speex to slin, cost 6
  == Registered translator 'lintospeex' from format slin to speex, cost 175
 [app_verbose.so] => (Send verbose output)
  == Registered application 'Verbose'
 [app_setcdruserfield.so] => (CDR user field apps)
  == Registered application 'SetCDRUserField'
  == Registered application 'AppendCDRUserField'
  == Manager registered action SetCDRUserField
 [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
  == Registered translator 'g726tolin' from format g726 to slin, cost 42
  == Registered translator 'lintog726' from format slin to g726, cost 103
 [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
  == Registered application 'LookupBlacklist'
 [app_getcpeid.so] => (Get ADSI CPE ID)
  == Registered application 'GetCPEID'
 [app_enumlookup.so] => (ENUM Lookup)
  == Registered application 'EnumLookup'
  == Parsing '/etc/asterisk/enum.conf': Found
 [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
  == Registered translator 'ilbctolin' from format ilbc to slin, cost 59
  == Registered translator 'lintoilbc' from format slin to ilbc, cost 112
 [pbx_config.so] => (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
    -- Registered extension context 'default'
    -- Including context 'capiin' in context 'default'
    -- Registered extension context 'capiin'
    -- Added extension '797302' priority 1 (CID match '0175xxxxxxx')to capiin
    -- Added extension '797302' priority 2 (CID match '0175xxxxxxx')to capiin
    -- Added extension '797302' priority 3 (CID match '0175xxxxxxx')to capiin
    -- Registered extension context 'capidialtone'
    -- Added extension 's' priority 1 to capidialtone
    -- Added extension 's' priority 2 to capidialtone
    -- Including context 'capiin' in context 'capidialtone'
    -- Registered extension context 'dialout'
    -- Added extension 's' priority 1 to dialout
    -- Added extension 's' priority 2 to dialout
    -- Added extension 's' priority 3 to dialout
    -- Added extension 's' priority 4 to dialout
    -- Added extension 's' priority 5 to dialout
    -- Added extension 's' priority 102 to dialout
    -- Added extension 'h' priority 1 to dialout
    -- Registered extension context 'dial_now'
    -- Added extension '_.' priority 1 to dial_now
    -- Added extension '_.' priority 2 to dial_now
    -- Added extension '_.' priority 3 to dial_now
 [app_segfault.so] => (Application for crashing Asterisk with a segmentation fault)
  == Registered application 'Segfault'
 [app_read.so] => (Read Variable Application)
  == Registered application 'Read'
 [app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
  == Parsing '/etc/asterisk/alarmreceiver.conf': Found
  == Registered application 'AlarmReceiver'
 [format_gsm.so] => (Raw GSM data)
  == Registered file format gsm, extension(s) gsm
 [app_dial.so] => (Dialing Application)
  == Registered application 'Dial'
 [app_striplsd.so] => (Strip trailing digits)
  == Registered application 'StripLSD'
 [app_capiECT.so] => ((CAPI*) ECT)
  == Registered application 'capiECT'
 [app_disa.so] => (DISA (Direct Inward System Access) Application)
  == Registered application 'DISA'
 [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call)
  == Registered application 'NoCDR'
 [app_image.so] => (Image Transmission Application)
  == Registered application 'SendImage'
 [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver)
 [app_cut.so] => (Cuts up variables)
  == Registered application 'Cut'
 [app_devstate.so] => (Application for sending device state messages)
  == Registered channel type 'DS' (Application for sending device state messages)
  == Manager registered action Devstate
  == Registered application 'Devstate'
 [app_festival.so] => (Simple Festival Interface)
  == Registered application 'Festival'
 [app_meetme.so] => (MeetMe conference bridge)
  == Registered application 'MeetMeAdmin'
  == Registered application 'MeetMeCount'
  == Registered application 'MeetMe'
 [app_echo.so] => (Simple Echo Application)
  == Registered application 'Echo'
 [chan_phone.so] => (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
 [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
  == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
 [app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
  == Registered application 'PrivacyManager'
 [app_flash.so] => (Flash zap trunk application)
  == Registered application 'Flash'
 [skipping app_intercom.so]
 [app_setcallerid.so] => (Set CallerID Application)
  == Registered application 'SetCallerPres'
  == Registered application 'SetCallerID'
 [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
 [app_capiHOLD.so] => ((CAPI*) HOLD)
  == Registered application 'capiHOLD'
 [app_substring.so] => ((Deprecated) Save substring digits in a given variable)
  == Registered application 'SubString'
 [chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
  == Parsing '/etc/asterisk/skinny.conf': Found
Jul 29 13:03:21 WARNING[10616]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
  == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny))
 [format_sln.so] => (Raw Signed Linear Audio support (SLN))
  == Registered file format sln, extension(s) sln|raw
 [app_queue.so] => (True Call Queueing)
  == Registered application 'Queue'
  == Manager registered action Queues
  == Manager registered action QueueStatus
  == Manager registered action QueueAdd
  == Manager registered action QueueRemove
  == Registered application 'AddQueueMember'
  == Registered application 'RemoveQueueMember'
  == Parsing '/etc/asterisk/queues.conf': Found
 [app_mp3.so] => (Silly MP3 Application)
  == Registered application 'MP3Player'
 [app_lookupcidname.so] => (Look up CallerID Name from local database)
  == Registered application 'LookupCIDName'
 [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
  == Registered file format wav, extension(s) wav
 [app_senddtmf.so] => (Send DTMF digits Application)
  == Registered application 'SendDTMF'
 [format_vox.so] => (Dialogic VOX (ADPCM) File Format)
  == Registered file format vox, extension(s) vox
 [app_waitforring.so] => (Waits until first ring after time)
  == Registered application 'WaitForRing'
 [app_setcidnum.so] => (Set CallerID Number)
  == Registered application 'SetCIDNum'
 [skipping chan_oss.so]
 [app_talkdetect.so] => (Playback with Talk Detection)
  == Registered application 'BackgroundDetect'
 [app_db.so] => (Database access functions for Asterisk extension logic)
  == Registered application 'DBget'
  == Registered application 'DBput'
  == Registered application 'DBdel'
  == Registered application 'DBdeltree'
 [chan_sip.so] => (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
  == SIP Listening on 0.0.0.0:5060
  == Using TOS bits 0
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 20000
Asterisk Ready.

Und nun die Ausgabe wenn ich ein Callback initiere und eine Nr wähle:
Code:
    -- creating pipe for PLCI=0x101 msn = 797302
       > sent ALERT_REQ PLCI = 0x101
    -- started pbx on channel (callgroup=0)!
    -- Executing Wait("CAPI[contr1/797302]/0", "1") in new stack
    -- Executing System("CAPI[contr1/797302]/0", "cp /etc/asterisk/callMichael /var/spool/asterisk/outgoing/") in new stack
    -- Executing Hangup("CAPI[contr1/797302]/0", "") in new stack
  == Spawn extension (capiin, 797302, 3) exited non-zero on 'CAPI[contr1/797302]/0'            => Ist ein "non-zero" Ende schlimm?
    -- CAPI Hangingup
       > sent CONNECT_RESP for PLCI = 0x101
    -- removed pipe for PLCI = 0x101
    -- Attempting call on CAPI/797302:07071xxxxxx for s@capidialtone:1 (Retry 1)
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x4
       > sent FACILITY_REQ (PLCI=0x101)
       > Channel CAPI[contr1/797302]/1 was answered.
    -- Executing Dial("CAPI[contr1/797302]/1", "CAPI/@797302:b") in new stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x7
    -- Called @797302:b
       > sent FACILITY_REQ (PLCI=0x201)
    -- CAPI[contr1/797302]/2 is making progress passing it to CAPI[contr1/797302]/1
       > sent CALLEDPARTYNUMBER INFO digit = 1 (PLCI=0x201)
       > sent CALLEDPARTYNUMBER INFO digit = 7 (PLCI=0x201)
       > sent CALLEDPARTYNUMBER INFO digit = 5 (PLCI=0x201)     => Hier verschluckt er Ziffern!!!
       > sent CALLEDPARTYNUMBER INFO digit = 2 (PLCI=0x201)
       > sent CALLEDPARTYNUMBER INFO digit = 2 (PLCI=0x201)
    -- CAPI Hangingup
       > sent DISCONNECT_B3_REQ NCCI=0x20201
       > sent DISCONNECT_REQ PLCI=0x201
    -- removed pipe for PLCI = 0x201
  == Spawn extension (capidialtone, s, 1) exited non-zero on 'CAPI[contr1/797302]/1'
    -- CAPI Hangingup
       > sent DISCONNECT_B3_REQ NCCI=0x10101
       > sent DISCONNECT_REQ PLCI=0x101
    -- removed pipe for PLCI = 0x101
Jul 29 13:06:30 NOTICE[10679]: pbx_spool.c:242 attempt_thread: Call completed to CAPI/797302:07071xxxxxx
 
Zuletzt bearbeitet:
Holla!

Was ich vergessen hab zu erwähnen, die ISDN Karte ist an eine Fritz!Box 7050 mit 14.4.06 Firmware angeschlossen. Kann es daran liegen dass die DTMF-Töne vom Handy "verfälscht" werden?
 
Hallo Quetsch,

bist du sicher das dein Provider deiner Phoneflat DTMF-Töne unterstützt? Ich hatte aus diesem Grunde bei meiner Callback Installation die gleichen Probleme.
Um dieses zu testen braucht du nur in deinem Callfile vor deiner Nummer, die zurückgerufen werden soll, die Telekom Vorwahl 01033 voranstellen. Viele Billig-Provider unterstützen DTMF leider nicht.

Salu2

Molto aus Spanien
 
Hi molto!

Ja, ich bin mir sicher, da das Callback wie gesagt über 1&1 bisher ja problemlos lief. Ich hab wegen einigen Problemen auf Suse 10.0 upgedatet und in einem Schwung auch die Fritz!Box daheim...
Ich tippe ja mittlerweile eher auf die Box. Leider kann ich tintins Tip erst nächste Woche ausprobieren, da sie bei meinen Eltern steht und ich armer Student sie nur per Callback "schmarotzerig" mitbenutze...
 
Der Schuldige ist tatsächlich die neue FW!

Hallo!

Der Schuldige ist identifiziert!

Es ist eindeutig die aktuelle Firmware der Fritz!Box!
Mit Version 14.04.06 läuft die DTMF Erkennung noch einwandfrei, mit allen anderen nicht mehr zuverlässig!
Die o.g. Aktivierung der Outband DTMF-Signalisierung brachte zwar eine Verbesserung, dennoch wurden bei einer 11-stelligen Nummer immer noch 1-2 Ziffern verschluckt, mit der 06er Version aber keine einzige!!!

Nun belasse ich die Box erstmal bei dieser Version, bis geklärt ist warum...

Will uns da AVM "ärgern", oder an welchen Parametern haben die da geschraubt, das DTMF nimmer so gut funzt??

Soweit so gut,

Quetsch.
 
Gibt es inzwischen eine Lösung für die neue Firmware?
 
Als Threadersteller habe ich mich nicht weiter mit dem Problem beschäftigt, da es nach dem Downgrade wieder funktionierte... Für Lösungsideen wäre ich auch dankbar... ;-)
 
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