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GMX und Grandstream 486

Dieses Thema im Forum "GMX" wurde erstellt von PPPawlo, 22 Feb. 2005.

  1. PPPawlo

    PPPawlo Neuer User

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    Ich komme ins Internet (hniter dem Modem), funktioniert als "Router" aber ich bekomme nur "besetzt". Hier ist meine Konfiguration. Kann das einer von euch erfahrenen Leuten evtl mal kurz durchchecken - wär ganz toll!! Per Netphone über Software/Computer funktioniert alles. 1000 Dank!


    Grandstream HandyTone 486 Configuration
    >
    Code:
    MAC Address: 	  00.0B.82.02.82.89
    Product Model: 	  HT487
    Software Version: 	  Program--1.0.5.16    Bootloader--1.0.0.21    HTML--1.0.0.41    VOC--1.0.0.7
      	  detected NAT type is open Internet
      	 
    Admin Password: 	   xxx(admin user password to configure this IP phone)
    End User Password: 	 xxx (end user password to configure this IP phone)
    WAN IP Address: vorhanden
    X dynamically assigned via DHCP (default) or PPPoE
    (will attempt PPPoE if DHCP fails and following is non-blank)
      	PPPoE account ID: XXXX	
      	PPPoE password: XXXX	
       Use this DNS server (if specified): leer. . .
    statically configured as:
           	IP Address: 	  . . .
           	Subnet Mask: 	  . . .
           	Default Router: 	  . . .
           	DNS Server 1: 	  . . .
           	DNS Server 2: 	  . . .
    SIP Server: sip-gmx.net	  (e.g., sip.mycompany.com, or IP address)
    Outbound Proxy: leer	  (e.g., proxy.myprovider.com, or IP address, if any)
    SIP User ID: 49XXX(TelNummer)	  (the user part of an SIP address)
    Authenticate ID: 49XXX(Telnummer)	  (can be identical to or different from SIP User ID)
    Authenticate Password: XXX 	 
    Name: 	  (optional, e.g., John Doe)leer
     
    Advanced Options: 	 
    Preferred Vocoder:
    (in listed order) 	  choice 1:  PCMU
      choice 2:  PCMA
      choice 3:  G723
      choice 4:  
      choice 5:  
      choice 6:  
      choice 7:  
    G723 rate: 	X  6.3kbps encoding rate       5.3kbps encoding rate
    iLBC frame size: X	  20ms       30ms
    iLBC payload type: 99	  (between 96 and 127, default is 98)
    Silence Suppression: 	X  No      Yes
    Voice Frames per TX: 2	  (up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
    Layer 3 QoS: 48	  (Diff-Serv or Precedence value)
    Layer 2 QoS: 	  802.1Q/VLAN Tag 0    802.1p priority value (0-7) 0
    Use DNS SRV: 	 X No      Yes
    User ID is phone number: 	  No     X Yes (habe mit beidem experimentiert)
    SIP Registration: 	  Yes     No X
    Unregister On Reboot: 	  Yes     No X
    Register Expiration: 	60  (in minutes. default 1 hour, max 45 days)
    Early Dial: 	 X No      Yes (use "Yes" only if proxy supports 484 response)
    Dial Plan Prefix: 	  (this prefix string is added to each dialed number)
    No Key Entry Timeout: 4	  (in seconds, default is 4 seconds)
    Use # as Dial Key: 	  No     X Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
    local SIP port: 	5060  (default 5060)
    local RTP port: 	5004  (1024-65535, default 5004)
    Use random port: 	X  No      Yes
    NAT Traversal: 	  No   
      X Yes, STUN server is: (URI or IP:port) stun.gmx.net
    keep-alive interval: 	20  (in seconds, default 20 seconds)
    Use NAT IP 	  (if specified, this IP address is used in SIP/SDP message)
    Proxy-Require: 	  (if specified, the content will appear in Proxy-Require header)
    TFTP Upgrade Server: 	  . . . (for remote software upgrade and configuration)
    HTTP Upgrade Server: 	  (IP address or URL)
    Auto Upgrade: 	 X No      Yes, check for new firmware every days (default 7 days)
    SUBSCRIBE for MWI: 	  No, do not send SUBSCRIBE for Message Waiting Indication
      Yes, send periodical SUBSCRIBE for Message Waiting Indication
    Offhook Auto-Dial: 	  (User ID/extension to dial automatically when offhook)
    Enable Call Features: 	  No      Yes (if Yes, Call Forwarding & Do-Not-Disturb are supported locally)
    Disable Call-Waiting: X	  No      Yes
    Send DTMF: 	   Xin-audio     via RTP (RFC2833)     via SIP INFO
    DTMF Payload Type: 101 
    Send Flash Event: 	X  No      Yes   (Flash will be sent as a DTMF event if set to Yes)
    FXS Impedance: 	 600
    NTP Server: 	time.nist.gov  (URI or IP address)
    Time Zone: 	 
    Daylight Savings Time: 	  No      Yes X  (if set to Yes, display time will be 1 hour ahead of normal time)
    Send Anonymous: 	  No X      Yes   (caller ID will be blocked if set to Yes)
    Lock keypad update: 	  No  X    Yes   (configuration update via keypad is disabled if set to Yes)
    WAN side http access: 	  No  X    Yes   (WAN side access to http server will be rejected if set to No)
     
    NAT/DHCP Server Information & Configuration:
    WAN IP Address: 	  0.0.0.0
    Cloned WAN MAC Addr: 	leer            (in hex format)
    LAN Subnet Mask: 	  (default is 255.255.255.0)
    LAN DHCP Base IP: 	  (base IP for the LAN port, default is 192.168.2.1)
    DHCP IP Lease Time: 	  (in units of hours, default is 120 hours or 5 days)
    DMZ IP: 	 
    Port Forwarding: 	  WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
      WAN port    LAN IP    LAN port    Protocol
    PSTN access code: 	leer  (key pattern to use the PSTN line, default is "*00")
     
  2. voodoobaby

    voodoobaby Guest

    sip registration auf yes
    send dtmf auf via sip info

    bei stun server habe ich : stun.gmx.net:3478
     
  3. Gast

    Gast Guest

    Meines Erachtens heißt der STUN bei GMX "stun.gmx.de",probiere es mal damit! Außerdem setze mal "Layer 3 QoS" auf 176 !
    Mal einen Versuch damit starten.
    Gruß von Tom
     
  4. PPPawlo

    PPPawlo Neuer User

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    Hat geklappt - Ihr seid einfach super! Danke!!