Hallo, ich habe gestern mein Cisco Phone 7940 eingerichtet mit 1&1 und Sipgate. Ich habe bei 1&1 den Komplett 16000 Anschluß und wollte eine von den 3 Telefonnummern, auf meinem Cisco nutzen. Ich habe die Ports 5060-5069, 5080-5089 und die Media Ports 16384-32766 freigegeben und weiterleitet auf das Telefon. Mit 1&1 kann ich raus telefonieren, mit Sipgate raus und rein telefonieren. Also würde ich fast sagen es liegt nicht an den Portweiterleitungen. Ich habe hier schon ein Post gefunden, welches das gleiche Problem beschreibt von 2005, doch gabs damals keine Lösung, vielleicht hat sich ja nun etwas geändert. http://www.ip-phone-forum.de/archive/index.php/t-83538.html Ich hab auch den Log einmal eingeschaltet, so wie es in dem oben genannten Post vorgeschlagen wurde und hier das Ergebnis. Was mir jetzt komisch vorkommt, ich habe mit dem Handy (Vodafone) versucht auf die 1&1 Rufnummer anzurufen, nur warum steht dann im Log immer was von Sigpgate? Hier meine SIPDefault Code: # Image Version image_version: "P0S3-07-2-00" # Proxy Server proxy1_address: "sip.1und1.de" # IP address here alternatively proxy2_address: "sipgate.de" # Proxy Server Port (default - 5060) proxy1_port:"5060" proxy2_port:"5060" # Emergency Proxy info proxy_emergency: "" # IP address here alternatively proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "1" nat_address: "xxxx.redirectme.net" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "1" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "3600" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "g711ulaw" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "1" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: "1" # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: "avt" ~np~# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: "3" # SIP Timers timer_t1: "500" ; Default 500 msec timer_t2: "4000" ; Default 4 sec sip_retx: "10" ; Default 11 sip_invite_retx: "6" ; Default 7 timer_invite_expires: "180" ; Default 180 sec # Setting for Message speeddial to UOne box messages_uri: "*97" # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "./" # Time Server sntp_mode: "unicast" sntp_server: "ntp.sipgate.net" # IP address here alternatively time_zone: "GMT" dst_offset: "2" dst_start_month: "April" dst_start_day: "" dst_start_day_of_week: "Sun" dst_start_week_of_month: "1" dst_start_time: "02" dst_stop_month: "Oct" dst_stop_day: "" dst_stop_day_of_week: "Sunday" dst_stop_week_of_month: "8" dst_stop_time: "2" dst_auto_adjust: "1" # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: "0" ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: "1" ; Default 1 (Call Waiting enabled) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: "101" ; Default 100 # XML file that specifies the dialplan desired dial_template: "dialplan" # Network Media Type (auto, full100, full10, half100, half10) network_media_type: "auto" #Autocompletion During Dial (0-off, 1-on [default]) autocomplete: "1" #Time Format (0-12hr, 1-24hr [default]) time_format_24hr: "1" # Remote Party ID remote_party_id: 1 ; 0-Disabled (default), 1-Enabled Und hier die SIP<MAC> Code: # SIP Configuration Generic File # Image Version image_version: P0S3-07-2-00 phone_label: "CISCO 7940 " # Line 1 appearance line1_displayname: "49231xxx" line1_shortname:"1&1" line1_name: 49231xxx line1_authname: "49231xxx" line1_password: "xxx" # Line 2 appearance line2_displayname: "12551xx" line2_shortname: "Sipgate" line2_name: 12551xx line2_authname: "12551xx" line2_password: "xx" # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: "xxxx" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none Ich hoffe mir kann jemand helfen, vielen Dank schon einmal für die Bemühungen. Grüße Tobias