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Problem Abgehender Ruf zu Sipgate ???

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von Enterprise, 5 Juni 2005.

  1. Enterprise

    Enterprise Neuer User

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    Hallo Forum,

    Ich habe Asterisk auf Kanotix laufen.
    Asterisk meldet sich bei Sipgate an.
    Ich kann Anrufe bekommen über Sipgate.
    Aber abgehen ist etwas faul.

    Hier habe ich die Meldung vom CLI Mode:

    Code:
       -- Executing Dial("SIP/77-8158", "SIP/06********@sipgate|60") in new stack
        -- Called 06*********@sipgate
    Jun  5 13:58:17 NOTICE[2653]: chan_sip.c:6875 handle_response: Failed to authenticate on INVITE to '"77" <sip:4318***@si
    pgate.net>;tag=as2eac11bf'
      == Spawn extension (default, *5206*********, 1) exited non-zero on 'SIP/77-8158'
        -- Got SIP response 481 "Call Leg Does Not Exist" back from 217.10.79.9
    Asterisk*CLI>
    
    Hier ist meine extension.cfg :

    Code:
    [general]
    
    static=yes
    writeprotect=yes
    
    [globals]
    
    
    [incoming-sipgate]
    exten => 4318***,1,Dial(SIP/77)
    
    
    [incoming-capi]  ;ankommend von CAPI
    exten => 9,1,SetAccount(isdn)
    exten => 9,2,Disa,no-password|default
    exten => 44,1,SetLAnguage(de)
    exten => 44,2,VoiceMAil(11)
    
    
    exten => 45,1,Wait(1)
    exten => 45,2,Answer()
    exten => 45,3,capiHOLD
    exten => 45,4,capiECT(${REMOTE-CAPI}:b11)
    ;exten => 45,3,Answer
    ;exten => 45,4,Playtones(440)
    exten => t,4,Playback(invalid)
    
    
    [default]
    include => intern
    include => kennziffern
    include => parkedcalls
    include => enum
    include => sipout
    ;---------------------------------------------------------------------
    
    kennziffern]
    exten => 3333,1,Goto(mainmenu,s,1)
    exten => 95,1,SetLanguage(de)
    exten => 95,2,VoicemailMain()
    exten => 95,3,Hangup()
    
    exten => 96,1,VoiceMail(11)
    
    exten => 900,1,MeetMe(|E)
    exten => 901,1,MeetMe(901|M)
    exten => 902,1,MeetMe(902|M)
    
    
    exten => 910,1,MeetMe(9999)
    exten => 911,1,Directory(default)
    exten => 912,1,SayAlpha(abcdefghijk)
    exten => 913,1,Wait(1)
    exten => 913,2,Read(SOUND_NO|custom/menu1|2)
    exten => 913,3,Record(custom/${SOUND_NO}:gsm)
    exten => 913,4,Goto(default,913,2)
    exten => 914,1,Read(SOUND_NO|custom/menu1|2)
    exten => 914,2,Playback(custom/${SOUND_NO})
    exten => 914,3,Goto(default,914,1)
    exten => 915,1,Playback(aufnahme)
    
    exten => 920,1,SetLanguage(de)
    exten => 920,2,Background(pbx-invalid)
    exten => 920,3,Goto(920,1)
    
    exten => 982,1,Goto(mainmenu,s,1)
    exten => 983,1,Goto(demo-1,s,1)
    
    exten => 990,1,SetLanguage(de)
    exten => 990,2,Answer
    exten => 990,3,Queue(gruppe1|Tt|||145)
    exten => 990,4,VoiceMail(u77)
    exten => 991,1,SetLanguage(de)
    exten => 991,2,Answer
    exten => 991,3,Queue(gruppe2||||145)
    exten => 991,4,VoiceMail(u77)
    
    exten => 998,1,AgentLogin()
    exten => 999,1,AgentCallbackLogin()
    
    ;-------- Extern Routing --------------------------------------
    exten => _0[2-9].,1,SetAccount(extern)
    exten => _0[2-9].,2,SetAMAFlags(billing)
    exten => _0[2-9].,3,Dial(SIP/${EXTEN}@sipsnip,30)
    exten => _0[2-9].,4,Dial(SIP/${EXTEN}@sipgate,30)
    exten => _0[2-9].,5,Dial(SIP/${EXTEN}@sipgate,30)
    exten => _0[2-9].,7,Playback(invalid)
    
    exten => _0[1].,1,SetAccount(mobile)
    exten => _0[1].,2,SetAMAFlags(billing)
    exten => _0[1].,3,Dial(SIP/${EXTEN}@freenet,30)
    exten => _0[1].,4,Dial(SIP/${EXTEN}@sipsnip,30)
    exten => _0[1].,5,Dial(SIP/${EXTEN}@sipgate,30)
    exten => _0[1].,6,Playback(invalid)
    
    exten => _00.,1,SetAccount(ausland)
    exten => _00.,2,SetAMAFlags(billing)
    exten => _00.,3,Dial(SIP/${EXTEN}@sipsnip,30)
    exten => _00.,4,Dial(SIP/${EXTEN}@sipgate,30)
    exten => _00.,5,Dial(SIP/${EXTEN}@sipsnip,30)
    exten => _00.,6,Playback(invalid)
    
    exten => _[1-9]XXX.,1,SetAccount(extern)
    exten => _[1-9]XXX.,2,SetAMAFlags(billing)
    exten => _[1-9]XXX.,3,Dial(SIP/06203${EXTEN}@sipsnip,30)
    exten => _[1-9]XXX.,4,Dial(SIP/06203${EXTEN}@freenet,30)
    exten => _[1-9]XXX.,5,Dial(SIP/06203${EXTEN}@sipgate,30)
    exten => _[1-9]XXX.,6,Playback(invalid
    
    exten => _*50.,1,Dial(SIP/${EXTEN:3}@sipsnip,60)
    exten => _*51.,1,Dial(SIP/${EXTEN:3}@freenet,60)
    exten => _*52.,1,Dial(SIP/${EXTEN:3}@sipgate,60)
    exten => _*53.,1,Dial(SIP/${EXTEN:3}@webde,60)
    exten => _*55.,1,Dial(Zap/g1/${EXTEN:3})
    
    exten => _*1XX,1,Dial(${REMOTE-SKC}/${EXTEN:2})
    ;exten => _[1-2]X,1,Dial(Zap/g1/${EXTEN})
    ;------------ Kurzwahl ---------------------------------------
    exten => 8000,1,Goto(kennziffern,0172********,1)
    
    exten => t,1,Playback(invalid)
    ;---------------------------------------------------------------
    
    [enum]
    exten => _*1.,1,EnumLookup(${EXTEN:2})
    exten => _*1.,2,Dial(${ENUM})
    
    
    [intern]
    exten => 77,1,Dial(SIP/77,15,rtT)
    exten => 77,2,Queue(gruppe2|r||15)
    ;exten => 77,2,VoiceMail(u77)
    
    exten => 78,1,Dial(SIP/78,15,rtT)
    exten => 78,2,Queue(gruppe1|r||15)
    exten => 78,3,VoiceMail(u78)
    
    exten => 11,1,Dial(Zap/g1/11)
    exten => 21,1,Dial(Zap/g1/21)
    
    
    [submenu]
    exten => 1,1,Goto(default,66,1)
    exten => 2,1,Goto(default,77,2)
    exten => s,1,Ringing()
    exten => s,2,Wait(2)
    exten => s,3,Background(men2)
    
    [mainmenu]
    exten => 1,1,Goto(submenu,s,1)
    exten => 2,1,Goto(demo,s,1)
    exten => 3,1,Dial(SIP/06*********}@freenet,30,tr)
    exten => 4,1,Hangup()
    exten => 5,1,Directory(default)
    exten => s,1,Ringing()
    exten => s,2,Wait(4)
    exten => s,3,Answer()
    exten => s,4,Background(men1)
    
    ;[sipout]
    ;exten => _*52.,1,SetCallerId,4318***
    ;exten => _*52.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)
    ;exten => _*52.,3,Hangup
    
    
    [demo-1]
    exten => #,1,Playback(demo-thanks)
    exten => #,2,Hangup()
    exten => 1000,1,Goto(default,s,1)
    exten => 1234,1,Playback(transfer,skip)
    exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
    exten => 1235,1,Voicemail(u1234)
    exten => 1236,1,Dial(Console/dsp)
    exten => 1236,2,Voicemail(u1234)
    exten => 2,1,BackGround(demo-moreinfo)
    exten => 2,2,Goto(s,6)
    exten => 3,1,SetLanguage(fr)
    exten => 3,2,Goto(s,5)
    exten => 500,1,Playback(demo-abouttotry)
    exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default)
    exten => 500,3,Playback(demo-nogo)
    exten => 500,4,Goto(s,6)
    exten => 600,1,Playback(demo-echotest)
    exten => 600,2,Echo()
    exten => 600,3,Playback(demo-echodone)
    exten => 600,4,Goto(s,6)
    exten => i,1,Playback(invalid)
    exten => s,1,Wait(1)
    exten => s,2,Answer()
    exten => s,3,DigitTimeout(5)
    exten => s,4,ResponseTimeout(10)
    exten => s,5,BackGround(demo-congrats)
    exten => s,6,BackGround(demo-instruct)
    exten => t,1,Goto(#,1)
    
    [macro-stdexten]
    exten => a,1,VoicemailMain(${ARG1})
    exten => s,1,Dial(${ARG2},20)
    exten => s,2,Goto(s-${DIALSTATUS},1)
    exten => s-.,1,Goto(s-NOANSWER,1)
    exten => s-BUSY,1,Voicemail(b${ARG1})
    exten => s-BUSY,2,Goto(default,s,1)
    exten => s-NOANSWER,1,Voicemail(u${ARG1})
    exten => s-NOANSWER,2,Goto(default,s,1)
    
    und hier meine sip.conf:

    Code:
    ; Useful CLI commands to check peers/users:
    ;   sip show peers		Show all SIP peers (including friends)
    ;   sip show users		Show all SIP users (including friends)
    ;   sip show registry		Show status of hosts we register with
    ;   sip debug			Show all SIP messages
    
    [general]
    context=default			; Default context for incoming calls
    ;recordhistory=yes		; Record SIP history by default
    				; (see sip history / sip no history)
    ;realm=mydomain.tld		; Realm for digest authentication
    				; defaults to "asterisk"
    				; Realms MUST be globally unique according to RFC 3261
    				; Set this to your host name or domain name
    port=5060			; UDP Port to bind to (SIP standard port is 5060)
    ;bindaddr=192.168.1.50		; IP address to bind to (0.0.0.0 binds to all)
    ;srvlookup=no			; Enable DNS SRV lookups on outbound calls
    				; Note: Asterisk only uses the first host
    				; in SRV records
    				; Disabling DNS SRV lookups disables the
    				; ability to place SIP calls based on domain
    				; names to some other SIP users on the Internet
    
    ;pedantic=yes			; Enable slow, pedantic checking for Pingtel
    				; and multiline formatted headers for strict
    				; SIP compatibility (defaults to "no")
    tos=0x18                        ; Set IP QoS to either a keyword or numeric val
    ;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none
    maxexpirey=12000		; Max length of incoming registration we allow
    defaultexpirey=12000		; Default length of incoming/outoing registration
    ;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
    ;videosupport=yes		; Turn on support for SIP video
    
    ;disallow=all			; First disallow all codecs
    ;allow=all
    allow=ulaw			; Allow codecs in order of preference
    ;allow=gsm			; Note: codec order is respected only in [general]
    musicclass=default		; Sets the default music on hold class for all SIP calls
    				; This may also be set for individual users/peers
    language=de			; Default language setting for all users/peers
    				; This may also be set for individual users/peers
    ;relaxdtmf=yes			; Relax dtmf handling
    ;rtptimeout=60			; Terminate call if 60 seconds of no RTP activity
    				; when we're not on hold
    ;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
    				; when we're on hold (must be > rtptimeout)
    ;trustrpid = no			; If Remote-Party-ID should be trusted
    ;progressinband=no		; If we should generate in-band ringing always
    ;useragent=Asterisk PBX		; Allows you to change the user agent string
    nat=yes				; NAT settings
                                    ; yes = Always ignore info and assume NAT
                                    ; no = Use NAT mode only according to RFC3581
                                    ; never = Never attempt NAT mode or RFC3581 support
    ;promiscredir = no      ; If yes, allows 302 or REDIR to non-local SIP address
    ;register => 1234:password@mysipprovider.com
    ;
    externip = exyz*****.no-ip.com	; Address that we're going to put in outbound SIP messages
    ;externip = 80.184.174.5	; if we're behind a NAT
    
    				; The externip and localnet is used
    				; when registering and communicating with other proxies
    				; that we're registered with
    				; You may add multiple local networks.  A reasonable set of defaults
    				; are:
    localnet=192.168.1.0/255.255.255.0; All RFC 1918 addresses are local networks
    ;canreinvite=no
    insecure=very
    dtmf=rfc2833
    
    ;-------------------------------------------------------------------------
    ; Sipgate Einstellungen
    ;-------------------------------------------------------------------------
    ;port = 5060
    bindaddr = 0.0.0.0
    ;context = sip-out
    qualify=no
    ;disable=all
    allow=alaw
    ;allow=ulaw
    ;allow=g729
    allow=gsm
    allow=slinear
    srvlookup=yes
    canreinvite=yes
    ;language=de
    ;register => SIPID:PASSWD@sipgate.de/SIPID
    
    ;[sipgate-out]
    ;type=friend
    ;insecure=very ; otherwise I get authentication errors
    ;nat=yes
    ;username=SIPID
    ;fromuser=SIPID
    ;fromdomain=sipgate.de
    ;secret=SIPPW
    ;host=sipgate.de
    ;qualify=yes
    
    ;-----------------------------------------------------------------------------------
    
    register=>4318***:xyzyxz@sipgate.de/4318***
    
    [sipgate]
    type=friend
    username=4318***
    secret=passwort
    host=sipgate.de
    fromuser=4318***
    fromdomain=sipgate.net
    context=incoming-sipgate
    canreinvite=no
    qualify=yes
    allow=alaw
    allow=ulaw
    ;allow=g729
    allow=gsm
    insecure=very
    nat=yes
    dtmf=rfc2833
    tos=0x18
    accountcode=sipgate
    
    ;[freenet]
    ;type=friend
    ;username=yxtzkk
    ;secret=password
    ;host=freenet.de
    ;fromuser=yxtzkk
    ;fromdomain=freenet.de
    ;context=incoming-freenet
    ;canreinvite=no
    ;qualify=yes
    ;allow=ulaw
    ;insecure=very
    ;nat=yes
    ;dtmf=rfc2833
    ;tos=0x18
    ;accountcode=freenet
    
    [77]
    type=friend
    username=77
    secret=77
    callerid="77" <77>
    host=dynamic
    nat=no                        ; X-Lite is behind a NAT router
    canreinvite=no                ; Typically set to NO if behind NAT
    ;disallow=all
    allow=alaw
    allow=ulaw
    ;allow=g729
    allow=gsm                     ; GSM consumes far less bandwidth than ulaw
    context=default
    mailbox=77
    
    [78]
    type=friend
    username=78
    secret=78
    callerid="78" <78>
    host=dynamic
    nat=no                        ; X-Lite is behind a NAT router
    canreinvite=no                ; Typically set to NO if behind NAT
    ;disallow=all
    allow=alaw
    allow=ulaw
    ;allow=g729
    allow=gsm                     ; GSM consumes far less bandwidth than ulaw
    context=default
    mailbox=78
    
    Dann habe ich noch ein zweites Problem,

    Wenn zwei Sipteilnehmer mit einander verbunden sind dann kommt aus dem Lautsprecher => chr chr chr chr chr was kann das sein ???

    mfg

    Enterprise
     
  2. betateilchen

    betateilchen Grandstream-Guru

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    am Letzenberg
    Ich würde sagen, da schnarcht einer ...

    [hr:d499f21868]

    Benenne mal Deinen Sipgate-Context um in die ID-Nummer. Hier ein Beispiel aus meiner SIP.conf, das problemlos funktioniert (bei mir taucht übrigens niemals sipgate.net auf)

    Code:
    
    register => 2635xxx:yyyyyy@sipgate.de/2635xxx
    
    [2635xxx]
    type=peer
    username=2635xxx
    fromuser=2635xxx
    secret=yyyyyyy
    context=default
    host=sipgate.de
    fromdomain=sipgate.de
    insecure=very
    caninvite=no
    canreinvite=no
    nat=no
    disallow=all
    allow=ulaw
    
    Im Dial steht dann (Sipgate wird über *34 verwendet)
    Code:
    exten => _*34.,1,SetCallerID(2635xxx) ; Sipgate möchte die IP gesetzt haben ...
    exten => _*34.,n,Dial(SIP/${EXTEN:3}@2635xxx,30,r)
    exten => _*34.,n,HangUp
    
     
  3. TinTin

    TinTin Aktives Mitglied

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    ...und in der sip.conf auf ein "qualify=no" achten im [sipgate] Eintrag.
     
  4. Maik

    Maik Gesperrt

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    Versuchs mal mit 'fromdomain=sipgate.de'.
     
  5. Enterprise

    Enterprise Neuer User

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    Danke an euch,

    Habe alles beachtet und es geht jetzt.

    mfg

    Enterprise