7940 und 1&1 - keine eingehende Anrufe

[tob1as1987]

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.

[10:49:06] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <217.10.79.9>:<5060>, handle = 8
[10:49:06] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <217.10.79.9>:<5060>, handle=<8>:message=<REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 77.178.123.84:5061;branch=z9hG4bK642f5e41 From: sip:[email protected] To: sip:[email protected] Call-ID: [email protected] Date: Mon, 15 Sep 2008 08:49:06 GMT CSeq: 279 REGISTERUser-Agent: CSCO/7 Contact: <sip:[email protected]:5061> Content-Length: 0 Expires: 3600 , length=<368>
[10:49:06] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8
[10:49:06] SIPProcessUDPMessage: recv UDP message from <217.10.79.9>:<50195>:<SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 77.178.123.xxx:5061;branch=z9hG4bK642f5e41 From: sip:[email protected] To: sip:[email protected];tag=e6312186bd5c2eb4571f6550243cfacc.c31c Call-ID: [email protected] CSeq: 279 REGISTER WWW-Authenticate: Digest realm="sipgate.de", nonce="48ce22ae1a066d0d9276abc4652f914e082330c0" Content-Length: 0>, length=383
[10:49:06] sipSPICheckResponse: Response match: [email protected], cseq=279, cseq_method=REGISTER
[10:49:06] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <217.10.79.9>:<5060>, handle = 8
[10:49:06] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <217.10.79.9>:<5060>, handle=<8>:message=<REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 77.178.123.xxx:5061;branch=z9hG4bK0f7c2b09 From: sip:[email protected] To: sip:[email protected] Call-ID: [email protected] Date: Mon, 15 Sep 2008 08:49:06 GMT CSeq: 280 REGISTER User-Agent: CSCO/7 Contact: <sip:[email protected]:5061> Authorization: Digest username="12551xx",realm="sipgate.de",uri="sip:sipgate.de" ,response="d2f0782d1b53d92fbc83d05839f7d788",nonce="48ce22ae1a066d0d9276abc4652f914e082330c0",algorithm=md5 Content-Length: 0 Expires: 3600>, length=<557>
[10:49:06] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8
[10:49:06] SIPProcessUDPMessage: recv UDP message from <217.10.79.9>:<50195>:<SIP/2.0 200 OK Via: SIP/2.0/UDP 77.178.123.xxx:5061;branch=z9hG4bK0f7c2b09 From: sip:[email protected] To: sip:[email protected];tag=e6312186bd5c2eb4571f6550243cfacc.62b9 Call-ID: [email protected] CSeq: 280 REGISTER Contact: <sip:[email protected]:5061>;expires=600 Content-Length: 0>, length=333
[10:49:06] sipSPICheckResponse: Response match: [email protected], cseq=280, cseq_method=REGISTER

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

# 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>

# 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