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Probleme mit Asterisk beim rauswählen

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von ululu, 12 Nov. 2006.

  1. ululu

    ululu Neuer User

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    Hi @ all!

    Bin schon eine weile am rumbasteln, schaffe es aber nicht das das ich über sipgate rauswählen kann. Intern geht alles einwandfrei und ich kann auch über die sipgatenummer rein wählen. Nur eben raus geht leider nicht. Ich poste mal die configs und das debug.

    sip.conf
    Code:
    [general]
    context=default			; Default context for incoming calls
    port=5060			; UDP Port to bind to (SIP standard port is 5060)
    bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
    srvlookup=yes			; 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
    				
    language=de			; Default language setting for all users/peers
    externip=xxxxx.mine.nu	; Address that we're going to put in outbound SIP messages
    localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
    
    ;qualify=no
    disable=all
    allow=alaw
    allow=ulaw
    allow=g729
    allow=gsm
    allow=slinear
    canreinvite=yes 
    
    register => SipID:SipPW@sipgate.at/SipID
    
    
    [SipID]
    type=peer
    username=SipID
    secret=SipPW
    host=sipgate.at
    fromuser=SipID
    fromdomain=sipgate.at
    insecure=very
    canreinvite=no
    nat=no
    ;disallow=all
    ;allow=ulaw
    context=abgehend
    
    [sipgate_at_in]
    type=peer
    fromdomain=sipgate.at
    host=sipgate.at
    context=ankommend
    
    [101]
    type=friend
    username=101
    secret=101
    callerid="101" <101>
    host=dynamic
    nat=no                       
    canreinvite=no                
    ;disallow=all
    ;allow=gsm                    
    allow=ulaw
    ;allow=alaw
    context=home
    dtmfmode=rfc2833
    
    [102]
    type=friend
    username=102
    secret=102
    callerid="102" <102>
    host=dynamic
    nat=no                      
    canreinvite=no                
    ;disallow=all
    ;allow=gsm                   
    allow=ulaw
    ;allow=alaw
    context=home
    dtmfmode=rfc2833
    
    extensions.conf
    Code:
    [general]
    static=yes
    writeprotect=no
    
    [globals]
    
    [default]
    
    ;exten => s,1,dial(SIP/101&SIP/102,8,t)
    
    include => home
    include => ankommend
    include => abgehend
    
    [home]
    exten => 101,1,Dial(SIP/101,30)
    exten => 101,2,Hangup
    
    exten => 102,1,Dial(SIP/102,30)
    exten => 102,3,Hangup
    
    
    [ankommend]
    exten => SipID,1,Dial(SIP/101)
    
    [abgehend]
    exten => _0.,1,Dial,SIP/${EXTEN}@SipID|45|r
    
    Und das passiert wenn ich vom Phone eine Nummer anrufe:

    Code:
    Sip read:
    ACK sip:TELEFONNUMMER@192.168.0.253 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK08e4f82f
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a51752002113a72296-17905334
    To: <sip:TELEFONNUMMER@192.168.0.253>;tag=as4547effc
    Call-ID: 0003e3a5-1752000f-20b964e1-0b9c6007@192.168.0.201
    CSeq: 102 ACK
    Content-Length: 0
    
    
    Urgent handler
    7 headers, 0 lines
    Urgent handler
    Destroying call '0003e3a5-1752000f-20b964e1-0b9c6007@192.168.0.201'
    Urgent handler
    
    
    Sip read:
    
    Urgent handler
    1 headers, 0 lines
    Urgent handler
    
    
    Sip read:
    
    Urgent handler
    1 headers, 0 lines
    Urgent handler
    
    
    Sip read:
    INVITE sip:TELEFONNUMMER@192.168.0.253 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    Max-Forwards: 70
    CSeq: 101 INVITE
    User-Agent: Cisco-CP7960G/8.0
    Contact: <sip:102@192.168.0.201:5060;transport=udp>
    Expires: 180
    Accept: application/sdp
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
    Remote-Party-ID: "102" <sip:102@192.168.0.201>;party=calling;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,norefersub
    Content-Length: 275
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    
    v=0
    o=Cisco-SIPUA 25461 0 IN IP4 192.168.0.201
    s=SIP Call
    t=0 0
    m=audio 22466 RTP/AVP 0 8 18 101
    c=IN IP4 192.168.0.201
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/0
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    
    Urgent handler
    17 headers, 13 lines
    Urgent handler
    Using latest request as basis request
    Urgent handler
    Sending to 192.168.0.201 : 5060 (non-NAT)
    Urgent handler
    Reliably Transmitting (no NAT):
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>;tag=as0cfc1e13
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    CSeq: 101 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:TELEFONNUMMER@192.168.0.253>
    Proxy-Authenticate: Digest realm="asterisk", nonce="67a6e27d"
    Content-Length: 0
    
    
     to 192.168.0.201:5060
    Urgent handler
    Scheduling destruction of call '0003e3a5-17520010-416a6bfc-46977836@192.168.0.201' in 15000 ms
    Urgent handler
    Found user '102'
    Urgent handler
    
    
    Sip read:
    ACK sip:TELEFONNUMMER@192.168.0.253 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK1372b47d
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>;tag=as0cfc1e13
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    CSeq: 101 ACK
    Content-Length: 0
    
    
    Urgent handler
    7 headers, 0 lines
    Urgent handler
    
    
    Sip read:
    INVITE sip:TELEFONNUMMER@192.168.0.253 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    Max-Forwards: 70
    CSeq: 102 INVITE
    User-Agent: Cisco-CP7960G/8.0
    Contact: <sip:102@192.168.0.201:5060;transport=udp>
    Proxy-Authorization: Digest username="102",realm="asterisk",uri="sip:TELEFONNUMMER@192.168.0.253",response="ccc3b065ffc966a04bbcfa522f262e8f",nonce="67a6e27d",algorithm=md5
    Expires: 180
    Accept: application/sdp
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
    Remote-Party-ID: "102" <sip:102@192.168.0.201>;party=calling;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,norefersub
    Content-Length: 275
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    
    v=0
    o=Cisco-SIPUA 25461 0 IN IP4 192.168.0.201
    s=SIP Call
    t=0 0
    m=audio 22466 RTP/AVP 0 8 18 101
    c=IN IP4 192.168.0.201
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/0
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    
    Urgent handler
    18 headers, 13 lines
    Urgent handler
    Using latest request as basis request
    Urgent handler
    Sending to 192.168.0.201 : 5060 (non-NAT)
    Urgent handler
    Found user '102'
    Urgent handler
    Found RTP audio format 0
    Urgent handler
    Found RTP audio format 8
    Urgent handler
    Found RTP audio format 18
    Urgent handler
    Found RTP audio format 101
    Urgent handler
    Peer audio RTP is at port 192.168.0.201:22466
    Urgent handler
    Found description format PCMU
    Urgent handler
    Found description format PCMA
    Urgent handler
    Found description format G729
    Urgent handler
    Found description format telephone-event
    Urgent handler
    Capabilities: us - 0x8014e (gsm|ulaw|alaw|slin|g729|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
    Urgent handler
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Urgent handler
    Looking for TELEFONNUMMER in home
    Urgent handler
    Reliably Transmitting (no NAT):
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>;tag=as0cfc1e13
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Contact: <sip:TELEFONNUMMER@192.168.0.253>
    Content-Length: 0
    
    
     to 192.168.0.201:5060
    Urgent handler
    
    
    Sip read:
    ACK sip:TELEFONNUMMER@192.168.0.253 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bK620a60c2
    From: "102" <sip:102@192.168.0.253>;tag=0003e3a5175200221209b25f-72bd2e97
    To: <sip:TELEFONNUMMER@192.168.0.253>;tag=as0cfc1e13
    Call-ID: 0003e3a5-17520010-416a6bfc-46977836@192.168.0.201
    CSeq: 102 ACK
    Content-Length: 0
    
    
    Urgent handler
    7 headers, 0 lines
    Urgent handler
    Destroying call '0003e3a5-17520010-416a6bfc-46977836@192.168.0.201'
    Urgent handler
    
    
    Sip read:
    
    Urgent handler
    1 headers, 0 lines
    Urgent handler
    
    Es ist sicherlich kein gewaltiger Fehler aber ich sehe ihn nicht mehr :) Danke im Vorraus

    LG
     
  2. stefanwillmerot

    stefanwillmerot Neuer User

    Registriert seit:
    6 Okt. 2006
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    Beruf:
    Softwareentwickler
    Ort:
    Amsterdam
    Mein Verdacht: Deine Telefone liegen im Kontext "home". Von dort können sie aber nicht rauswählen weil der Dial-Befehl im Kontext "abgehend" nicht erreichbar ist. Einige Fehlermeldungen würdest Du wahrscheinlich auf der Asterisk-Konsole sehen können.

    Pack den Dial-Befehl doch mal ins "home".

    Grüße
    Stefan
     
  3. ululu

    ululu Neuer User

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    Hey Cool :) Das war es! Danke!

    LG