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[Erledigt]Brauch Hilfe bei Konfiguration von SIP -> Asterisk -> misdn -> Isdn Telefon

Dieses Thema im Forum "Asterisk ISDN mit mISDN" wurde erstellt von MP_LEO, 3 Jan. 2009.

  1. MP_LEO

    MP_LEO Neuer User

    Registriert seit:
    12 Juni 2008
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    Ich glaube ich brauch nochmal euere Hilfe. Es geht um folgendes:

    Erstmal funktioniert momentan die Verbindung zwischen ISDN Telefon und Asterisk.

    Damit mir die Konfiguration leichter geht habe ich mir Asterisk-GUI installiert und der funktioniert auch schon soweit. Dort habe ich unter Trunks meine beiden SIP Account eingetragen, die sich auch zu meinem Provider verbinden. Wenn ich jetzt eine der beiden SIP Accounts mit meinem Handy anrufe kommen die auch bis zu meinem Asterisk Server. Nur leider wird mir dann gleich gesagt das der Teilnehmer den Anruf ablehnt (Steht zumindest auf dem Handy). Jetzt habe ich noch einen USER 6000 mit einer SIP Telefonnummer angelegt und will eigentlich das dieser User das ISDN Telefon verwendet. Nur leider geht das irgendwie nicht. Auch bei dem ISDN Telefon ist das problem wenn ich keine Regel erstelle für Telefonate die vom ISDN rein kommen das mir das Telefon sag kein Teilnehmer. Wenn ich haber jetzt eine Regel erstelle für das ISDN Telefon das der Anruf bei drücken der 0 die MailBox abfragen soll geht es. Was mach ich falsch wie kann ich vom ISDN Telefon über ein SIP Account telefonieren und wie kann ich es einstellen das von einem SIP Account die Telefonate an ein ISDN Telefon weitergeleitet werden.

    Die Datein wurden am anfang mit sampel erstell darum steht da soviel drin. Die kommentar habe ich mal lieber entfernt.

    sip.conf
    Code:
    [general]
    context = default
    allowoverlap = no
    bindport = 5060
    bindaddr = 192.168.100.100
    srvlookup = yes
    allowexternaldomains = yes
    allowguest = yes
    allowsubscribe = yes
    allowtransfer = yes
    alwaysauthreject = no
    autodomain = no
    callevents = no
    canreinvite = nonat
    checkmwi = 10
    compactheaders = no
    defaultexpiry = 120
    domain = lan
    dtmfmode = rfc2833
    dumphistory = no
    externrefresh = 10
    fromdomain = sip.qsc.de
    g726nonstandard = no
    jbenable = no
    jbforce = no
    jbimpl = 
    jblog = no
    jbmaxsize = 
    jbresyncthreshold = 
    language = 
    maxcallbitrate = 384
    maxexpiry = 3600
    minexpiry = 60
    mohinterpret = default
    mohsuggest = 
    nat = yes
    notifyringing = yes
    pedantic = yes
    progressinband = never
    promiscredir = no
    realm = asterisk
    recordhistory = no
    register = user[:secret[:authuser]]@host[:port][/contact]
    registerattempts = 0
    registertimeout = 20
    relaxdtmf = no
    rtpholdtimeout = 300
    rtptimeout = 60
    sendrpid = no
    sipdebug = yes
    subscribecontext = 
    t1min = 100
    t38pt_udptl = yes
    tos_audio = none
    tos_sip = none
    tos_video = none
    trustrpid = no
    useragent = Asterisk PBX
    usereqphone = no
    videosupport = no
    disallow = all
    allow = undefined,ulaw,alaw,gsm
    [authentication]
    users.conf
    Code:
    [general]
    
    fullname = New User
    
    userbase = 6000
    
    hasvoicemail = yes
    
    vmsecret = 1234
    
    hassip = yes
    
    hasiax = yes
    
    hasmanager = no
    
    callwaiting = yes
    threewaycalling = yes
    callwaitingcallerid = yes
    transfer = yes
    canpark = yes
    cancallforward = yes
    callreturn = yes
    callgroup = 1
    pickupgroup = 1
    vmexten = 1111
    
    [12349999990]
    context = DID_12349999990
    host = sip.qsc.de
    trunkname = Firmentelefon  ; GUI metadata
    username = 12349999990
    secret = password
    hasiax = no
    registeriax = no
    hassip = yes
    registersip = yes
    trunkstyle = voip
    hasexten = no
    fromdomain = sip.qsc.de
    fromuser = 12349999990
    insecure = port,invite
    authuser = 12349999990
    disallow = all
    allow = ulaw,alaw,gsm,g726
    
    [6000]
    username = 6000
    transfer = yes
    mailbox = 6000
    call-limit = 100
    fullname = 12349999990
    registersip = no
    host = dynamic
    callgroup = 1
    context = DLPN_DialPlan1
    cid_number = 6000
    hasvoicemail = yes
    vmsecret = 1234
    email = 
    threewaycalling = no
    hasdirectory = no
    callwaiting = no
    hasmanager = no
    hasagent = no
    hassip = yes
    hasiax = no
    secret = 
    nat = yes
    canreinvite = yes
    dtmfmode = rfc2833
    insecure = no
    pickupgroup = 1
    autoprov = yes
    label = 6000
    macaddress = :
    linenumber = 1
    LINEKEYS = 1
    disallow = all
    allow = ulaw,gsm
    
    [trunk_1]
    context = DID_trunk_1
    host = sip.qsc.de
    username = 12349999991
    secret = password
    trunkname = FAX  ; GUI metadata
    hasiax = no
    registeriax = no
    hassip = yes
    registersip = yes
    trunkstyle = voip
    hasexten = no
    insecure = no
    fromdomain = sip.qsc.de
    fromuser = 12349999991
    disallow = all
    allow = ulaw,alaw,gsm,g726
    
    [6001]
    username = 6001
    transfer = yes
    mailbox = 6001
    call-limit = 100
    fullname = 12349999991
    registersip = no
    host = dynamic
    callgroup = 1
    context = DLPN_DialPlan1
    cid_number = 6001
    hasvoicemail = no
    vmsecret = 
    email = 
    threewaycalling = no
    hasdirectory = no
    callwaiting = no
    hasmanager = no
    hasagent = no
    hassip = no
    hasiax = yes
    secret = 
    nat = yes
    canreinvite = no
    dtmfmode = rfc2833
    insecure = no
    pickupgroup = 1
    disallow = all
    allow = ulaw,gsm
    autoprov = no
    label = 
    macaddress = 
    linenumber = 1
    LINEKEYS = 1
    
    [trunk_2]
    context = DID_trunk_2
    host = 127.0.0.1
    username = iaxmodem
    secret = password
    trunkname = IAX-Modem  ; GUI metadata
    hasiax = yes
    registeriax = yes
    hassip = no
    registersip = no
    trunkstyle = voip
    hasexten = no
    disallow = all
    allow = all
    misdn.conf
    Code:
    [general]
    misdn_init = /etc/misdn-init.conf
    debug = 0
    ntdebugflags = 0
    ntdebugfile = /var/log/misdn-nt.log
    ntkeepcalls = no
    bridging = no
    l1watcher_timeout = 0
    stop_tone_after_first_digit = yes
    append_digits2exten = yes
    dynamic_crypt = no
    crypt_prefix = **
    crypt_keys = test,muh
    [default]
    context = misdn
    language = en
    musicclass = default
    senddtmf = yes
    far_alerting = no
    allowed_bearers = all
    nationalprefix = 0
    internationalprefix = 00
    rxgain = 0
    txgain = 0
    te_choose_channel = no
    pmp_l1_check = no
    reject_cause = 16
    need_more_infos = no
    nttimeout = no
    method = standard
    overlapdial = yes
    dialplan = 0
    localdialplan = 0
    cpndialplan = 0
    early_bconnect = yes
    incoming_early_audio = no
    nodialtone = no
    presentation = -1
    screen = -1
    echotraining = no
    jitterbuffer = 4000
    jitterbuffer_upper_threshold = 0
    hdlc = no
    max_incoming = -1
    max_outgoing = -1
    [intern]
    ports = 1,2
    context = Intern
    [internPP]
    ports = 3
    [first_extern]
    ports = 4
    context = Extern1
    msns = *
    [second_extern]
    ports = 5
    context = Extern2
    callerid = 15
    msns = 102,144,101,104
    [trunk_m1]
    trunkname = FirmentelefonISDN
    context = DID_trunk_m1
    ports = 1
    hasmisdn = yes
    msns = *
    extensions.conf
    Code:
    [general]
    static = yes
    writeprotect = no
    clearglobalvars = no
    [globals]
    CONSOLE = Console/dsp  ; Console interface for demo
    IAXINFO = guest  ; IAXtel username/password
    TRUNK = Zap/G2  ; Trunk interface
    TRUNKMSD = 1  ; MSD digits to strip (usually 1 or 0)
    FEATURES = 
    DIALOPTIONS = tThHkK
    RINGTIME = 20
    FOLLOWMEOPTIONS = 
    PAGING_HEADER = Intercom
    PAGING_TIMEOUT = 60
    12349999990 = SIP/12349999990
    CID_12349999990 = 12349999990
    CID_6000 = 12349999990
    trunk_m1 = mISDN/g:trunk_m1
    trunk_1 = SIP/trunk_1
    CID_trunk_1 = 12349999991
    CID_6001 = 12349999991
    trunk_2 = IAX2/trunk_2
    [dundi-e164-canonical]
    [dundi-e164-customers]
    [dundi-e164-via-pstn]
    [dundi-e164-local]
    include => dundi-e164-canonical
    include => dundi-e164-customers
    include => dundi-e164-via-pstn
    
    [dundi-e164-switch]
    switch => DUNDi/e164
    
    [dundi-e164-lookup]
    include => dundi-e164-local
    include => dundi-e164-switch
    
    [macro-dundi-e164]
    exten => s,1,Goto(${ARG1},1)
    include => dundi-e164-lookup
    
    [iaxtel700]
    exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
    
    [iaxprovider]
    [trunkint]
    exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
    exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    
    [trunkld]
    exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
    exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    
    [trunklocal]
    exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    
    [trunktollfree]
    exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    
    [international]
    ignorepat => 9
    include => longdistance
    include => trunkint
    
    [longdistance]
    ignorepat => 9
    include => local
    include => trunkld
    
    [local]
    ignorepat => 9
    include => default
    include => trunklocal
    include => iaxtel700
    include => trunktollfree
    include => iaxprovider
    include => parkedcalls
    
    [macro-trunkdial]
    exten => s,1,Dial(${ARG1})
    exten => s,n,Goto(s-${DIALSTATUS},1)
    exten => s-NOANSWER,1,Hangup
    exten => s-BUSY,1,Hangup
    exten => _s-.,1,NoOp
    
    [macro-stdPrivacyexten];
    exten => s,1,Dial(${ARG2},20|p)  ; Ring the interface, 20 seconds maximum, call screening 
    exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to voicemail w/ unavail announce
    exten => s-NOANSWER,2,Goto(default,s,1)  ; If they press #, return to start
    exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy announce
    exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
    exten => s-DONTCALL,1,Goto(${ARG3},s,1)  ; Callee chose to send this call to a polite "Don't call again" script.
    exten => s-TORTURE,1,Goto(${ARG4},s,1)  ; Callee chose to send this call to a telemarketer torture script.
    exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user into VoicemailMain
    
    [macro-page];
    exten => s,1,ChanIsAvail(${ARG1}|js)  ; j is for Jump and s is for ANY call
    exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
    exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the PolyComs
    exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the Grandstream, Snoms, and Others
    exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!
    exten => s,n,Dial(${ARG1}||)
    exten => s,n(fail),Hangup
    
    [demo]
    exten => s,1,Wait(1)  ; Wait a second, just for fun
    exten => s,n,Answer  ; Answer the line
    exten => s,n,Set(TIMEOUT(digit)=5)  ; Set Digit Timeout to 5 seconds
    exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
    exten => s,n(restart),BackGround(demo-congrats)  ; Play a congratulatory message
    exten => s,n(instruct),BackGround(demo-instruct)  ; Play some instructions
    exten => s,n,WaitExten  ; Wait for an extension to be dialed.
    exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
    exten => 2,n,Goto(s,instruct)
    exten => 3,1,Set(LANGUAGE()=fr)  ; Set language to french
    exten => 3,n,Goto(s,restart)  ; Start with the congratulations
    exten => 1000,1,Goto(default,s,1)
    exten => 1234,1,Playback(transfer,skip)  ; "Please hold while..." 
    exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
    exten => 1235,1,Voicemail(1234,u)  ; Right to voicemail
    exten => 1236,1,Dial(Console/dsp)  ; Ring forever
    exten => 1236,n,Voicemail(1234,b)  ; Unless busy
    exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
    exten => #,n,Hangup  ; Hang them up.
    exten => t,1,Goto(#,1)  ; If they take too long, give up
    exten => i,1,Playback(invalid)  ; "That's not valid, try again"
    exten => 500,1,Playback(demo-abouttotry)  ; Let them know what's going on
    exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)  ; Call the Asterisk demo
    exten => 500,n,Playback(demo-nogo)  ; Couldn't connect to the demo site
    exten => 500,n,Goto(s,6)  ; Return to the start over message.
    exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
    exten => 600,n,Echo  ; Do the echo test
    exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
    exten => 600,n,Goto(s,6)  ; Start over
    exten => 76245,1,Macro(page,SIP/Grandstream1)
    exten => _7XXX,1,Macro(page,SIP/${EXTEN})
    exten => 7999,1,Set(TIMEOUT(absolute)=60)
    exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
    exten => 8500,1,VoicemailMain
    exten => 8500,n,Goto(s,6)
    
    [page]
    exten => _X.,1,Macro(page,SIP/${EXTEN})
    
    [default]
    exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default)
    exten = _#6XXX,n,VoiceMail(${MBOX})
    exten = a,1,VoicemailMain(${MBOX})
    exten = 1111,1,VoiceMailMain(${CALLERID(num)}@default)
    
    [macro-stdexten-followme]
    exten = s,1,Answer
    exten = s,2,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
    exten = s,3,Set(__FMCIDNUM=${CALLERID(num)})
    exten = s,4,Set(__FMCIDNAME=${CALLERID(name)})
    exten = s,5,Followme(${ARG1},${FOLLOWMEOPTIONS})
    exten = s,6,Voicemail(${ARG1},u)
    exten = s-NOANSWER,1,Voicemail(${ARG1},u)
    exten = s-BUSY,1,Voicemail(${ARG1},b)
    exten = s-BUSY,2,Goto(default,s,1)
    exten = _s-.,1,Goto(s-NOANSWER,1)
    exten = a,1,VoicemailMain(${ARG1})
    
    [conferences]
    
    [ringgroups]
    exten = 6400,1,Goto(ringroups-custom-1,s,1)
    
    [queues]
    exten = 6500,1,Queue(${EXTEN})
    
    [voicemenus]
    
    [voicemailgroups]
    
    [directory]
    
    [page_an_extension]
    
    [pagegroups]
    
    [asterisk_guitools]
    exten = executecommand,1,System(${command})
    exten = executecommand,n,Hangup()
    exten = record_vmenu,1,Answer
    exten = record_vmenu,n,Playback(vm-intro)
    exten = record_vmenu,n,Record(${var1})
    exten = record_vmenu,n,Playback(vm-saved)
    exten = record_vmenu,n,Playback(vm-goodbye)
    exten = record_vmenu,n,Hangup
    exten = play_file,1,Answer
    exten = play_file,n,Playback(${var1})
    exten = play_file,n,Hangup
    
    [macro-trunkdial-failover-0.3]
    exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1)
    exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)
    exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}:)})
    exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1)
    exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})})
    exten = s,n,Goto(1-dial,1)
    exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
    exten = 1-setgbobname,n,Goto(s,3)
    exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
    exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
    exten = 1-fmsetcid,n,Goto(1-dial,1)
    exten = 1-dial,1,Dial(${ARG1})
    exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1)
    exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
    exten = 1-CHANUNAVAIL,n,Hangup()
    exten = 1-CONGESTION,1,Dial(${ARG2})
    exten = 1-CONGESTION,n,Hangup()
    exten = 1-out,1,Hangup()
    
    [DID_12349999990]
    include = DID_12349999990_default
    
    [DID_12349999990_default]
    exten = _X,1,Goto(ringroups-custom-1,s,1)
    
    [macro-stdexten]
    exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
    exten = s,2,GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?5:3)
    exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
    exten = s,4,Goto(s-${DIALSTATUS},1)
    exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
    exten = s-NOANSWER,1,Voicemail(${ARG1},u)
    exten = s-NOANSWER,2,Goto(default,s,1)
    exten = s-BUSY,1,Voicemail(${ARG1},b)
    exten = s-BUSY,2,Goto(default,s,1)
    exten = _s-.,1,Goto(s-NOANSWER,1)
    exten = a,1,VoicemailMain(${ARG1})
    
    [macro-pagingintercom]
    exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
    exten = s,2,Page(${ARG1}|${ARG2})
    exten = s,3,Hangup
    
    [CallingRule_Firmentelefon]
    exten = _i,1,Macro(trunkdial-failover-0.3,${trunk_m1}/${EXTEN:0},,trunk_m1,trunk_m1)
    
    [DLPN_DialPlan1]
    include = CallingRule_Firmentelefon
    include = default
    include = parkedcalls
    include = conferences
    include = ringgroups
    include = voicemenus
    include = queues
    include = voicemailgroups
    include = directory
    include = pagegroups
    include = page_an_extension
    
    [DID_trunk_m1]
    include = DID_trunk_m1_default
    
    [DID_trunk_1]
    include = DID_trunk_1_default
    
    [DID_trunk_1_default]
    exten = 12349999990,1,Voicemail(6000,u)
    
    [DID_trunk_2]
    include = DID_trunk_2_default
    
    [DID_trunk_2_default]
    exten = _0,1,Goto(default,6001,1)
    
    [DID_trunk_m1_default]
    
    [CallingRule_FAX]
    exten = _X,1,Macro(trunkdial-failover-0.3,${trunk_2}/${EXTEN:0},,trunk_2,)
    
    [ringroups-custom-1]
    exten = s,1,NoOp(Firmentelefon)
    exten = s,n,Dial(SIP/6000,30,${DIALOPTIONS}i)
    exten = s,n,Voicemail(6000,u)
     
  2. MP_LEO

    MP_LEO Neuer User

    Registriert seit:
    12 Juni 2008
    Beiträge:
    27
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    hat siche erstmal erledigt bin auf FreePBX umgestiegen