Callthrough geht nicht mehr

joerg100

Neuer User
Mitglied seit
23 Sep 2006
Beiträge
44
Punkte für Reaktionen
0
Punkte
0
Hallo,

nachdem die Callthrough Funktion monatelang beste Dienste geleistet hatte, funktioniert sie seit heute nicht mehr.

Bei einem "reload" werden die folgenden Meldungen ausgegeben (im unten stehenden Code werden die Anrufe über 1234567 auf callthrough gelegt):

Code:
(none)*CLI> reload
  == Parsing '/etc/asterisk/cdr.conf': Found
  == Parsing '/etc/asterisk/dnsmgr.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 9078 -> 9097
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
Asterisk Queue Logger restarted
    -- Reloading module 'res_features.so' (Call Features Resource)
  == Parsing '/etc/asterisk/features.conf': Found
    -- Added extension '700' priority 1 to parkedcalls
    -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_alaw: using generic PLC
    -- Reloading module 'app_playback.so' (Sound File Playback Application)
    -- Reloading module 'chan_capi.so' (Common ISDN API Driver (1.0.2))
[Jun 27 09:13:44] WARNING[2157]: chan_capi.c:5797 reload: config reload is not s
upported yet.
    -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
  == Parsing '/etc/asterisk/iax.conf': Found
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
    -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))

 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found

    -- Reloading module 'codec_g726.so' (ITU G.726-32kbps G726 Transcoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_g726: using generic PLC
    -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_gsm: using generic PLC
    -- Reloading module 'codec_ulaw.so' (mu-Law Coder/Decoder)
  == Parsing '/etc/asterisk/codecs.conf': Found
    -- codec_ulaw: using generic PLC
    -- Reloading module 'pbx_config.so' (Text Extension Configuration)
  == Parsing '/etc/asterisk/extensions.conf': Found
  == Setting global variable 'CAPI_CALLERID' to 9999999'
  == Setting global variable 'MAILER_TO' to '[email protected]'
  == Setting global variable 'MAILER_FROM' to '[email protected]'
  == Setting global variable 'MAILER_SMTP' to 'smtp.du.de'
  == Setting global variable 'MAILER_USER' to '[email protected]'
  == Setting global variable 'MAILER_PASSWORD' to 'secret'
    -- Registered extension context 'macro-entfernegitter'
    -- Added extension 's' priority 1 to macro-entfernegitter
    -- Registered extension context 'headsettest'
    -- Added extension '_95!' priority 1 to headsettest
    -- Registered extension context 'callbacktest'
    -- Registered extension context 'callthroughtest'
    -- Added extension '_93!' priority 1 to callthroughtest
    -- Registered extension context 'callthroughauth'
    -- Added extension '_1234567!' priority 1 to callthroughauth
    -- Added extension 's' priority 1 to callthroughauth
    -- Added extension 's' priority 2 to callthroughauth
    -- Added extension 's' priority 3 to callthroughauth
    -- Added extension 's' priority 4 to callthroughauth
    -- Added extension '_93!' priority 105 to callthroughauth
    -- Added extension '_93!' priority 106 to callthroughauth
    -- Added extension 's' priority 5 to callthroughauth
    -- Added extension 'h' priority 1 to callthroughauth
    -- Registered extension context 'callthrough'
    -- Added extension 's' priority 1 to callthrough
    -- Added extension 's' priority 2 to callthrough
    -- Added extension 's' priority 3 to callthrough
    -- Added extension 's' priority 4 to callthrough
    -- Added extension '_X' priority 1 to callthrough
    -- Added extension '_X' priority 2 to callthrough
    -- Added extension '*' priority 1 to callthrough
    -- Added extension '#' priority 1 to callthrough
    -- Added extension '#' priority 2 to callthrough
    -- Added extension '#' priority 3 to callthrough
    -- Added extension '#' priority 4 to callthrough
    -- Added extension '#' priority 5 to callthrough
    -- Added extension '#' priority 6 to callthrough
[Jun 27 09:13:44] WARNING[2157]: pbx.c:4689 add_pri: Unable to register extensio
n '#', priority 6 in 'callthrough', already in use
    -- Added extension '#' priority 7 to callthrough
    -- Added extension 't' priority 1 to callthrough
    -- Added extension 't' priority 2 to callthrough
    -- Added extension 't' priority 3 to callthrough
    -- Added extension 'h' priority 1 to callthrough
    -- Registered extension context 'mohtest'
 
...

    -- Registered extension context 'festnetz_out'
    -- Registered extension context 'sip1_out'
    -- Added extension '_X.' priority 1 to sip1_out
    -- Added extension '_00X.' priority 102 to sip1_out
    -- Registered extension context 'sip2_out'
    -- Added extension '_X.' priority 1 to sip2_out
    -- Added extension '_00X.' priority 102 to sip2_out
    -- Registered extension context 'sip3_out'
    -- Added extension '_X.' priority 1 to sip3_out
    -- Added extension '_00X.' priority 102 to sip3_out
    -- Registered extension context 'sip4_out'
    -- Added extension '_X.' priority 1 to sip4_out
    -- Added extension '_00X.' priority 102 to sip4_out
    -- Registered extension context 'default'
    -- Including context 'lokal' in context 'default'
    -- Registered extension context 'sip771'
    -- Including context 'lokal' in context 'sip771'
    -- Including context 'sip1_out' in context 'sip771'
    -- Registered extension context 'sip772'
    -- Including context 'lokal' in context 'sip772'
    -- Including context 'festnetz_out' in context 'sip772'
    -- Including context 'sip2_out' in context 'sip772'
    -- Registered extension context 'sip773'
    -- Including context 'lokal' in context 'sip773'
    -- Including context 'festnetz_out' in context 'sip773'
    -- Including context 'sip3_out' in context 'sip773'
    -- Registered extension context 'sip774'
    -- Including context 'lokal' in context 'sip774'
    -- Including context 'festnetz_out' in context 'sip774'
    -- Including context 'sip2_out' in context 'sip774'
    -- Registered extension context 'sip775'
    -- Including context 'lokal' in context 'sip775'
    -- Including context 'festnetz_out' in context 'sip775'
    -- Including context 'sip1_out' in context 'sip775'
    -- Registered extension context 'sip776'
    -- Including context 'lokal' in context 'sip776'
    -- Including context 'festnetz_out' in context 'sip776'
    -- Including context 'sip3_out' in context 'sip776'
   ...

    -- Registered extension context 'capi_in1'
    -- Added extension '7777777' priority 1 to capi_in1
    -- Added extension '6666666' priority 1 to capi_in1
    -- Added extension '1234567' priority 1 to capi_in1
[Jun 27 09:13:45] WARNING[2157]: pbx.c:4689 add_pri: Unable to register extensio
n '1234567', priority 1 in 'capi_in1', already in use
[Jun 27 09:13:45] WARNING[2157]: pbx.c:4689 add_pri: Unable to register extensio
n '1234567', priority 1 in 'capi_in1', already in use
    -- Added extension '6666666' priority 1 to capi_in1
    -- Registered extension context 'capi_in3'
    -- Added extension '_9[1-8]!' priority 1 to capi_in3
    -- Added extension '_9[1-8]!' priority 2 to capi_in3
    -- Added extension '_90!' priority 1 to capi_in3
    -- Added extension '_90!' priority 2 to capi_in3
    -- Added extension '_99!' priority 3 to capi_in3

    -- Registered extension context 'mobile_in'
    -- Added extension 's' priority 1 to mobile_in
    -- Added extension 's' priority 2 to mobile_in
    -- Added extension 's' priority 3 to mobile_in
    -- Registered extension context 'sip_in'
    -- Added extension '1' priority 1 to sip_in
    -- Added extension '2' priority 1 to sip_in
    -- Added extension '3 priority 1 to sip_in
    -- Added extension '4' priority 1 to sip_in

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

Vielen Dank für Eure Hilfe!
 
Hallo,

meintest Du den Callthrough Teil in der Extension.conf:

Code:
;Callthrough testen
[callthroughtest]
; Prompt caller to authenticate and validate passcode
exten => _93!,1,Goto(callthroughauth,s,1)

; nicht per include in lokal aufzunehmen; wird ueber callthroughtest aufgerufen.
[callthroughauth]
; Prompt caller to authenticate and validate passcode
exten => _1234567!,1,goto(s,1) 
exten => s,1,answer
exten => s,2,Noop(${CALLERID(num)})
exten => s,3,wait,1
exten => s,4,authenticate,/etc/asterisk/passwd|j

; Log failed authentication and hangup
exten => _93!,105,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authentication Failed!" >> /var/log/asterisk/callthrough.log
exten => _93!,n,HangUp()

exten => s,5,Goto(callthrough,s,1)

; Log when call has been aborted
exten => h,1,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call has been aborted at ${STRFTIME(||%Y-%m-%d_%H-%M-%S)}"  >> /var/log/asterisk/callthrough.log

; nicht per include in lokal aufzunehmen; wird ueber callthroughtest aufgerufen.
[callthrough]
; Prompt caller to key-in number to be dialed and to finish with #
exten => s,1,Set(NR=)
exten => s,2,Background(vm-enter-num-to-call)
exten => s,3,Set(TIMEOUT(response)=30)
exten => s,4,WaitExten
exten => _X,1,Set(NR=${NR}${EXTEN})
exten => _X,2,Goto(s,3)

exten => *,1,Goto(s,1)

exten => #,1,NoOp(${NR})
exten => #,2,Playback(vm-dialout)
exten => #,3,NoOP(Dialing Now)
exten => #,4,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authenticated & dialing ${NR}"  >> /var/log/asterisk/callthrough.log
exten => #,5,NoOp(Dialing ${NR})
;exten => #,6,Dial,CAPI/ISDN1/${NR}|55|bd (Alte Version, Wahl über CAPI)
exten => #,6,Dial,SIP/${NR}@PROVIDER1|55|bd
exten => #,6,Dial,SIP/${NR}|55|bd
exten => #,n,HangUp()

; Log when connection timed out
exten => t,1,Busy(3)
exten => t,n,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call to ${NR} timed out."  >> /var/log/asterisk/callthrough.log
exten => t,n,HangUp()

; Log when call has finished
exten => h,1,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Call to ${NR} completed."  >> /var/log/asterisk/callthrough.log

Vielen Dank!
 
@joerg
In Deiner extensions Konfiguration hast Du einige Doppeleinträge, die Du erstmal korrigieren solltest.
Code:
exten => #,1,NoOp(${NR})
exten => #,2,Playback(vm-dialout)
exten => #,3,NoOP(Dialing Now)
exten => #,4,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authenticated & dialing ${NR}"  >> /var/log/asterisk/callthrough.log
exten => #,5,NoOp(Dialing ${NR})
;exten => #,6,Dial,CAPI/ISDN1/${NR}|55|bd (Alte Version, Wahl über CAPI)
exten => #,[COLOR=red][B]6[/B][/COLOR],Dial,SIP/${NR}@PROVIDER1|55|bd
[B][COLOR=seagreen]exten => #,[COLOR=red]6[/COLOR],Dial,SIP/${NR}|55|bd
[/COLOR][/B]exten => #,n,HangUp()
Am einfachsten ist es, wenn Du die Durchnummerierung dem Asterisk überlässt, indem Du nach der 1 einfach "n" einsetzt:
Code:
exten => #,1,NoOp(${NR})
exten => #,n,Playback(vm-dialout)
exten => #,n,NoOP(Dialing Now)
exten => #,n,system,chroot /oldroot echo "${STRFTIME(||%Y-%m-%d_%H-%M-%S)} - ${CALLERID(num)}: Authenticated & dialing ${NR}"  >> /var/log/asterisk/callthrough.log
...
Ausserdem mach der grün markierte Eintrag so keinen Sinn und sollte m.E. gelöscht werden.
Einen ähnlichen Fehler vermute ich in Deinem [capi_in1] Context.

Gruß
dynamic
 
Holen Sie sich 3CX - völlig kostenlos!
Verbinden Sie Ihr Team und Ihre Kunden Telefonie Livechat Videokonferenzen

Gehostet oder selbst-verwaltet. Für bis zu 10 Nutzer dauerhaft kostenlos. Keine Kreditkartendetails erforderlich. Ohne Risiko testen.

3CX
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.