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Konfigurationsdateien gesucht

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von mazi, 25 Mai 2005.

  1. mazi

    mazi Neuer User

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    Hallo,

    kann mich derzeit bei Sipgate anmelden. Doch ein Gespräch zwischen einem IAX SW-Client zu einem Teilnehmer der über meinen VoIP Provider (Sipgate) telefoniert konnte ich nicht vermitteln.

    Es wäre super, wenn jemand seine sip.conf, extension.conf und evtl. iax.conf posten könnte, welche ich als Grundlage für meine Konfiguration verwenden kann.

    Bereits im Voraus vielen Dank für Eure Beträge.

    Viele Grüße

    Mazi

    extensions.conf:

    Code:
    [general] 
    
    static=yes 
    
    writeprotect=yes 
    
    [globals] 
    
    
    [macro-stdiax] 
    
    exten => s,1,Dial(IAX2/${ARG1}|20|Ttr) 
    exten => s,2,Voicemail2(u${ARG2}) 
    exten => s,3,Hangup 
    exten => s,102,Voicemail2(b${ARG2}) 
    exten => s,103,Hangup 
    
    [fullaccess] 
    
    include => local 
    
    [local] 
    
    exten => 222,1,Macro(stdiax,michael,${EXTEN}) 
    
    [default]
    exten => ,1,Dial(IAX/222,60)
    
    sip.conf:
    Code:
    [general]
    port=5060
    bindaddr=0.0.0.0
    contex=sip-out
    qualify=no
    disable=all
    allow=gsm
    allow=ulaw
    allow=alaw
    allow=g729
    allow=gsm
    allow=slinear
    srvlookup=yes
    language=de
    register => 2021047:SIP-Passwort@sipgate.de/2021047
    
    [sip-out]
    type=friend
    insecure=very
    nmat=yes
    username=2021047
    fromuser=2021047
    fromdomain=sipgate.de
    secret=SIP-Passwort
    host=sipgate.de
    qualify=no
    

    iax.conf:

    Code:
    [general] 
    
    bindaddr=0.0.0.0
    bindport=4569
    context=noaccess 
    group=1 
    callgroup=1 
    pickupgroup=1 
    amaflags=default 
    bandwidth=high     ; changed form low to high 
    allow=all          ; changed from disallow to allow 
    allow=ulaw         ; changed from disallow to allow 
    allow=alaw         ; changed from disallow to allow 
    allow=gsm
    allow=iLBC
    allow=Speex 
    jitterbuffer=yes 
    dropcount=2 
    maxjitterbuffer=500 
    maxexccessbuffer=400 
    tos=throughput 
    mailboxdetail=yes
    
     
    
    [guest] 
    
    type=user 
    context=iaxguest 
    callerid="Guest IAX User" 
     
    
    [michael] 
    
    type=friend 
    username=michael 
    secret=password 
    auth=md5 
    host=dynamic 
    context=fullaccess 
    mailbox=222
    callerid="Michael"<222> 
    
     
  2. Netview

    Netview IPPF-Promi

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    Westerwald
    Könntest du dies bitte nochmals editieren und die config-dateien
    mit (code)...(/code) einstellen - geht leichter zu lesen und man muss nix downloaden ;-)

    PS: die runde Klammern durch eckige ersetzen!
     
  3. mazi

    mazi Neuer User

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    Nach dem ich die sip.conf und extensions.conf geändert habe, kann ich über Sipgate eine Verbindung zu meinem SW Phone aufbauen. Leider erfolgt aber keine Tonvermittlung.
    Hat jemand eine Idee, was ich machen muss?


    sip.conf

    Code:
    [general]
    port=5060
    bindaddr=0.0.0.0
    qualify=no
    disable=all
    allow=gsm
    allow=ulaw
    allow=alaw
    allow=g729
    allow=gsm
    allow=slinear
    srvlookup=yes
    language=de
    register => 2021047:EKEQ5C@sipgate.de/2021047
    
    [sipgate]
    type=peer
    context=incoming_sipgate
    insecure=very
    username=2021047
    fromuser=2021047
    authuser=2021047
    fromdomain=sipgate.de
    secret=EKEQ5C
    host=sipgate.de
    qualify=yes
    

    extensions.conf:

    Code:
    [general]
    
    static=yes
    
    writeprotect=yes
    
    
    
    [globals]
    
    [incoming_sipgate]
    exten => 2021047,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
    ;exten => 2021047,2,Dial(IAX2/{222}|20|Trt)
    exten => 2021047,2,Macro(stdiax,michael,222);
    exten => 2021047,3,Answer
    exten => 2021047,4,Wait,1
    exten => 2021047,5,Voicemail(su4)
    exten => 2021047,6,Hangup
    
    [macro-stdiax]
    
    exten => s,1,Dial(IAX2/${ARG1}|20|Ttr)
    exten => s,2,Voicemail2(u${ARG2})
    exten => s,3,Hangup
    exten => s,102,Voicemail2(b${ARG2})
    exten => s,103,Hangup
    
    
    
    [fullaccess]
    
    include => local
    
    
    [local]
    
    exten => 999,1,Answer;
    exten => 999,2,Background(demo-congrats)
    exten => 999,3,Queue(holdloop)
    exten => 999,4,Hangup
    
    exten => 111,1,Macro(stdiax,diaa,§{EXTEN})
    
    exten => 222,1,Macro(stdiax,michael,${EXTEN})
    
    exten => 333,1,Macro(stdiax,martin,${EXTEN})
    
    exten => _XXX.,1,Dial(CAPI/@01835991825021::${EXTEN})
    
    
    exten => s,1,Ring
    exten => s,2,Wait(5)
    exten => s,3,Answer
    exten => s,4,Playback(ss-noservice) ; invalid extension
    exten => s,5,Hangup
    
    [from-capi]
    exten => s,1,Wait,1
    exten => s,2,Dial(IAX2/222,60)
    exten => s,3,Hangup
    
    [default]
    exten => ,1,Dial(IAX/222,60)
    
     
  4. betateilchen

    betateilchen Grandstream-Guru

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    Beschränke Dich mal zum Testen, bis alles läuft, auf einen einzelnen Codec, den Du in allen Konfigurationen verwendest.

    Es heißt übrigens "disallow=all" - nicht "disable=all"
     
  5. TinTin

    TinTin Aktives Mitglied

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    Läuft Dein Asterisk hinter einem Router? Dann fehlen noch Einträge wie

    externip=MeineAdresse.dyndns.org
    localnet=192.168.0.0/255.255.255.0
    nat=yes

    im [general] context der sip.conf. dyndns adresse kostenlos bei dyndns.org besorgen und diese auch imRouter eintragen. localnet auf Dein Netz anpassen. Fern heißt es disallow=all nicht disable=all (Es sei denn das ist neu und ich kenne es noch nicht)

    im [sipgate] context qualify=no setzen und evtl auch noch ein nat=yes, mal testen dort mit dem ein oder anderen, gibt hier welche da klappts mit yes und andere schwören auf no

    Außerdem noch ports 10000-20000 (siehe rtp.conf) sowie port 5060 auf dem Router einstellen so daß weitergeleitet wird auf Asterisk

    Wenn's dann immer noch nicht geht melde Dich nochmal :)
     
  6. betateilchen

    betateilchen Grandstream-Guru

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    am Letzenberg
    und benenne mal den [sipgate] um in[2021047]
     
  7. mazi

    mazi Neuer User

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    Vielen Dank für die Info's. Habe Eure Verschläge bis auf den Eintrag der dyndns Adresse in meiner Konfiguration übernommen. Bzgl. der dyndns Adresse möchte ich nochmals mit unserem Admin sprechen, ob bereits eine Vorhanden ist und werde diese in die sip.conf entsprechend übernehmen.

    Tschüß

    Mazi
     
  8. betateilchen

    betateilchen Grandstream-Guru

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    ja und ? Funktioniert es denn jetzt ?
     
  9. mazi

    mazi Neuer User

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    Hallo,

    habe temporär von unserem Admin alle Ports freigeschaltet bekommen. Leider funktioniert es nicht. Es kommt derzeit keine Verbindung zustande.


    Asterisk output:
    Code:
    p54B1A50D:/etc/asterisk # asterisk -vvvvvvvvvc
      == Parsing '/etc/asterisk/asterisk.conf': Found
      == Parsing '/etc/asterisk/extconfig.conf': Found
    Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
    Written by Mark Spencer <markster@digium.com>
    =========================================================================
      == 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 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
     [res_musiconhold.so] => (Music On Hold Resource)
      == Parsing '/etc/asterisk/musiconhold.conf': Found
      == Registered application 'MusicOnHold'
      == Registered application 'WaitMusicOnHold'
      == Registered application 'SetMusicOnHold'
     [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(01835991825021,01835991825021,from-capi,0,1) (1,2,64)
        -- listening on contr1 CIPmask = 0x1fff03ff
      == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.5) aLaw CVS HEAD)
     [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
     [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'
     [skipping chan_modem_i4l.so]
     [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
    May 30 13:33:25 WARNING[2357]: chan_mgcp.c:4044 reload_config: Unable to get our IP address, MGCP disabled
      == 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'
     [skipping app_txtcidname.so]
     [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 1
     [chan_local.so] => (Local Proxy Channel)
      == Registered channel type 'Local' (Local Proxy Channel Driver)
     [skipping app_groupcount.so]
     [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 3
      == Registered translator 'lintolpc10' from format slin to lpc10, cost 6
     [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 1
      == Registered translator 'lintogsm' from format slin to gsm, cost 3
     [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))
      == Manager registered action IAXpeers
      == Parsing '/etc/asterisk/iax.conf': Found
      == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
      == Using TOS bits 8
      == 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'
     [skipping chan_agent.so]
     [skipping app_controlplayback.so]
     [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_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 2
      == Registered translator 'lintog726' from format slin to g726, cost 2
     [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 3
      == Registered translator 'lintoilbc' from format slin to ilbc, cost 18
     [pbx_config.so] => (Text Extension Configuration)
      == Parsing '/etc/asterisk/extensions.conf': Found
        -- Registered extension context 'incoming_sipgate'
        -- Added extension '2021047' priority 1 to incoming_sipgate
        -- Added extension '2021047' priority 2 to incoming_sipgate
        -- Added extension '2021047' priority 3 to incoming_sipgate
        -- Added extension '2021047' priority 4 to incoming_sipgate
        -- Added extension '2021047' priority 5 to incoming_sipgate
        -- Added extension '2021047' priority 6 to incoming_sipgate
        -- Registered extension context 'macro-stdiax'
        -- Added extension 's' priority 1 to macro-stdiax
        -- Added extension 's' priority 2 to macro-stdiax
        -- Added extension 's' priority 3 to macro-stdiax
        -- Added extension 's' priority 102 to macro-stdiax
        -- Added extension 's' priority 103 to macro-stdiax
        -- Registered extension context 'fullaccess'
        -- Including context 'local' in context 'fullaccess'
        -- Registered extension context 'local'
        -- Added extension '999' priority 1 to local
        -- Added extension '999' priority 2 to local
        -- Added extension '999' priority 3 to local
        -- Added extension '999' priority 4 to local
        -- Added extension '111' priority 1 to local
        -- Added extension '222' priority 1 to local
        -- Added extension '333' priority 1 to local
        -- Added extension '_XXX.' priority 1 to local
        -- Added extension 's' priority 1 to local
        -- Added extension 's' priority 2 to local
        -- Added extension 's' priority 3 to local
        -- Added extension 's' priority 4 to local
        -- Added extension 's' priority 5 to local
        -- Registered extension context 'from-capi'
        -- Added extension 's' priority 1 to from-capi
        -- Added extension 's' priority 2 to from-capi
        -- Added extension 's' priority 3 to from-capi
        -- Registered extension context 'default'
        -- Added extension '' priority 1 to default
     [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'
     [skipping chan_modem_bestdata.so]
     [app_cut.so] => (Cuts up variables)
      == Registered application 'Cut'
     [format_mp3.so] => (MP3 format [Any rate but 8000hz mono optimal])
      == Registered file format mp3, extension(s) mp3
     [skipping chan_modem.so]
     [app_festival.so] => (Simple Festival Interface)
      == Registered application 'Festival'
     [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'
     [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
    May 30 13:33:25 WARNING[2357]: chan_skinny.c:2587 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_zapateller.so] => (Block Telemarketers with Special Information Tone)
      == Registered application 'Zapateller'
     [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
     [skipping chan_modem_aopen.so]
     [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'
        -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
      == 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.
    *CLI>     -- Executing NoOp("SIP/2021047-a087", "--- "06926548617" <06926548617> calling on Sipgate (2021047) ---") in new stack
        -- Executing Macro("SIP/2021047-a087", "stdiax|michael|222") in new stack
        -- Executing Dial("SIP/2021047-a087", "IAX2/michael|20|Ttr") in new stack
    May 30 13:33:46 NOTICE[2373]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2'
      == Everyone is busy/congested at this time
        -- Executing VoiceMail2("SIP/2021047-a087", "b222") in new stack
    May 30 13:33:46 WARNING[2373]: app_voicemail.c:1545 leave_voicemail: No entry in voicemail config file for '222'
        -- Executing Hangup("SIP/2021047-a087", "") in new stack
      == Spawn extension (macro-stdiax, s, 103) exited non-zero on 'SIP/2021047-a087' in macro 'stdiax'
      == Spawn extension (incoming_sipgate, 2021047, 2) exited non-zero on 'SIP/2021047-a087'
        -- Registered 'michael' (AUTHENTICATED) at 192.168.168.69:4569
        -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
        -- Registered 'michael' (AUTHENTICATED) at 192.168.168.69:4569
        -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
        -- Registered 'michael' (AUTHENTICATED) at 192.168.168.69:4569
        -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
    
    *CLI> stop now
    

    sip.conf.

    Code:
    [general]
    port=5060
    bindaddr=0.0.0.0
    qualify=no
    disallow=all
    allow=gsm
    allow=ulaw
    allow=alaw
    allow=g729
    allow=gsm
    allow=slinear
    srvlookup=yes
    language=de
    externip=trz-telematik.dyndns.org
    localnet=192.168.168.191/255.255.255.0
    nat=yes
    register => 2021047:SIP-Passwort@sipgate.de/2021047
    
    [2021047]
    type=peer
    context=incoming_sipgate
    insecure=very
    username=2021047
    fromuser=2021047
    authuser=2021047
    fromdomain=sipgate.de
    secret=SIP-Passwort
    host=sipgate.de
    qualify=yes
    
    extensions.conf:

    Code:
    [general]
    
    static=yes
    
    writeprotect=yes
    
    
    
    [globals]
    
    [incoming_sipgate]
    exten => 2021047,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
    ;exten => 2021047,2,Dial(IAX2/{222}|20|Trt)
    exten => 2021047,2,Macro(stdiax,michael,222);
    exten => 2021047,3,Answer
    exten => 2021047,4,Wait,1
    exten => 2021047,5,Voicemail(su4)
    exten => 2021047,6,Hangup
    
    [macro-stdiax]
    
    exten => s,1,Dial(IAX2/${ARG1}|20|Ttr)
    exten => s,2,Voicemail2(u${ARG2})
    exten => s,3,Hangup
    exten => s,102,Voicemail2(b${ARG2})
    exten => s,103,Hangup
    
    
    
    [fullaccess]
    
    include => local
    
    
    [local]
    
    exten => 999,1,Answer;
    exten => 999,2,Background(demo-congrats)
    exten => 999,3,Queue(holdloop)
    exten => 999,4,Hangup
    
    exten => 111,1,Macro(stdiax,diaa,§{EXTEN})
    
    exten => 222,1,Macro(stdiax,michael,${EXTEN})
    
    exten => 333,1,Macro(stdiax,martin,${EXTEN})
    
    exten => _XXX.,1,Dial(CAPI/@01835991825021::${EXTEN})
    
    
    exten => s,1,Ring
    exten => s,2,Wait(5)
    exten => s,3,Answer
    exten => s,4,Playback(ss-noservice) ; invalid extension
    exten => s,5,Hangup
    
    [from-capi]
    exten => s,1,Wait,1
    exten => s,2,Dial(IAX2/222,60)
    exten => s,3,Hangup
    
    [default]
    exten => ,1,Dial(IAX/222,60)
    
     
  10. mazi

    mazi Neuer User

    Registriert seit:
    5 Apr. 2005
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    Hallo,

    es funktioniert doch. Mein Linux Server hatte eine dynamische IP Adresse, die diesmal nicht mehr der 192.168.168.191 entsprach. Nachdem dem Server die statische IP Adresse 192.168.168.191 zugewiesen habe, funktioniert der Rufaufbau und die Kommunikationvom Festnetz auf mein SIP Telefon.

    Gemeinsam mit unserem Systemadministrator werden einen Trace fahren, welche Ports wir für die Kommuniokation zwischen SIP-Provider und meinem IAX-Client über Asterisk benötigen, da aus Sicherheitsgründen unser Admin nicht bereit ist sämtliche Ports von 10000 bis 20000 auf dem Router freizugeben.

    Nochmals vielen Dank für Eure kompetente und schnelle Hilfe.

    Viele Grüße

    Mazi
     
  11. knitter

    knitter Neuer User

    Registriert seit:
    22 Apr. 2005
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    Normalerweise müsste 5060 UDP reichen