7940 an FritzBox 7490

[Patrix2911]

Hallo,

ich traue es mir fast gar nicht zu sagen bzw. zu fragen. Ich habe mir ... vor einigen Jahren ein 7940 von Cisco zugelegt, hin und wieder mal probiert das an meiner FritzBox zum laufen zu bringen... leider immer ohne Erfolg, da es nicht Lebenswichtig war hab ich es dann einfach immer wieder in den Schrank geworfen. Nun würde ich es aber doch gern endlich mal benutzen und an meiner FritzBox 7490 betreiben.

Die Konfiguration wird auch über TFTP geladen, aber scheinbar hat das Telefon Probleme sich an der FritzBox anzumelden. Auf dem Display neben den beiden Leitungen ist ein Telefon mit einem X zu sehen. Nehme ich den Hörer ab, Höre ich ein Amtszeichen, aber das Wählen von Telefonnummern erfolgt nicht.

Es wäre schön, wenn mein 6ter Versuch dieses Telefon zu benutzen nun endlich gelingen würde.

Hier mal die Konfiguration, vielleicht findet ja jemand von euch den Fehler

Spoiler: SIPDefault.cnf

# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-8-12-00

# Call Manager
call_manager1_addr: ""
call_manager1_sip_port: 5060
call_manager2_addr: ""
call_manager2_sip_port: 5060
call_manager3_addr: ""
call_manager3_sip_port: 5060
call_manager4_addr: ""
call_manager4_sip_port: 5060
call_manager5_addr: ""
call_manager5_sip_port: 5060
call_manager6_addr: ""
call_manager6_sip_port: 5060

# Proxy Server
proxy1_address: ""
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""

# Proxy Server Port
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 500

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw

# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5

# 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

# 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 10
sip_invite_retx: 6             ; Default 6
timer_invite_expires: 180     ; Default 180 sec

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ""

# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.53.103.104"    ; SNTP Server IP Address
sntp_mode: unicast                ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET                    ; Time Zone Phone is in
dst_offset: 1                    ; Offset from Phone's time when DST is in effect
dst_start_month: April            ; Month in which DST starts
dst_start_day: ""                ; Day of month in which DST starts
dst_start_day_of_week: Sun        ; Day of week in which DST starts
dst_start_week_of_month: 1        ; Week of month in which DST starts
dst_start_time: 02                ; Time of day in which DST starts
dst_stop_month: Oct                ; Month in which DST stops
dst_stop_day: ""                ; Day of month in which DST stops
dst_stop_day_of_week: Sunday    ; Day of week in which DST stops
dst_stop_week_of_month: 8        ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2                ; Time of day in which DST stops
dst_auto_adjust: 1                ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1                ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

# 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0        ; Default 0 (Disable blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101        ; Default 101

# Sync value of the phone used for remote reset
sync: 1                        ; Default 1

# Backup Proxy Support
proxy_backup: ""            ; Dotted IP of Backup Proxy
proxy_backup_port: 5060        ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: ""         ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060    ; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0                ; VAD setting 0-disable (Default), 1-enable

# NAT/Firewall Traversal
nat_enable: 0               ; 0-Disabled (default), 1-Enabled
nat_address: ""                ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060     ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384     ; Start RTP range for media (default - 16384)
end_media_port: 32766       ; End RTP range for media (default - 32766)
nat_received_processing: 0    ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""             ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060   ; default is 5060

# 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: 1    ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 1                ; 0-Disabled (default), 1-Enabled, 2-Privileged

# XML URLs
services_url: ""            ; URL for external Phone Services
directory_url: ""            ; URL for external Directory location
logo_url: ""                ; URL for branding logo to be used on phone display

# HTTP Proxy Support
http_proxy_addr: ""            ; Address of HTTP Proxy server
http_proxy_port: 80            ; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""            ; restricted to dotted IP
dyn_dns_addr_2: ""            ; restricted to dotted IP
dyn_tftp_addr: ""            ; restricted to dotted IP

# Remote Party ID
remote_party_id: 0            ; 0-Disabled (default), 1-Enabled

# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0        ; Default 0 (Call Hold Ringback feature is off)

# Dialtone Stutter for MWI
stutter_msg_waiting: 0        ; 0-Disabled (default), 1-Enabled

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0                ; 0-Disabled (default), 1-Enabled

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

Spoiler: SIP-MAC-ADRESSE-.cnf

# SIP Configuration Generic File

# Call Manager
call_manager1_addr: "192.168.178.1"
call_manager1_sip_port: 5060
call_manager2_addr: "192.168.178.1"
call_manager2_sip_port: 5060

# Line 1
proxy1_address: "USECALLMANAGER"
proxy1_port: 5060
line1_name: "621"
line1_authname: "12345678"
line1_password: "Abcdefgh1981"
line1_displayname: "Mein Name"
line1_shortname: "Kurzname"

# Line 2
proxy2_address: "USECALLMANAGER"
proxy2_port: 5060
line2_name: "620"
line2_authname: "teltextil"
line2_password: "MeinPasswort!"
line2_displayname: "Mein Name 2e"
line2_shortname: "Kurzname 2"

phone_label: "Telefon"

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "Mein Telefon"

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco"
[CODE]
[/SPOILER]

Ich hoffe jemand von euch findet den kleinen entscheidenden Fehler.

 

[koyaanisqatsi]

Moinsen


Wenn der Benutzername des ersten angelegten IP-Telefons in der FRITZ!Box teltextil ist, dann gehört der zu allererst auch als Name in der Config.

Also, anstatt...

line1_name: "620"
line1_authname: "teltextil"

...dann...

line1_name: "teltextil"
line1_authname: "teltextil"

 

[Patrix2911]

Beide Benutzernamen sind in der FritzBox angelegt, auch "Abcdefgh" ...

 

[koyaanisqatsi]

Nunja, ich schrieb ja dass es die der IP-Telefone sein müssen.
Also...
Telefonie > Telefoniegeräte
...dann Stiftsymbol eines IP-Telefons klicken...
>Tab> Anmeldedaten.
Diese werden in deiner .cnf als Anmeldedaten benötigt.

 

[Patrix2911]

Ja, in der Fritzbox sind 2 WLAN/LAN Telefone angelegt.

Gerät 1: Ip1, User: 12345678 Pass: Abcdefgh1981 Intern: **621
Gerät 2: Ip2, User: teltextil Pass: MeinPasswort! Intern: **620

Das sind auch die Daten die in der CNF Datei hinterlegt sind.

 

[koyaanisqatsi]

Also hast du...
line1_name: "621"
...und...
line2_name: "620"
...schon auf diese (User) Namen geändert und obige .CNF ist eine Uralte?

In...
line1_displayname: "Mein Name"
...und...
line2_displayname: "Mein Name"
...kannste reinschreiben was du willst.

 

[Patrix2911]

Hallo, sorry - ich kam gestern leider gar nicht dazu mich mit dem Ding weiter zu beschäftigen. :/

Hier mal der aktuelle Stand (eigentlich ist alles unverändert) ich habe nochmal alle "Geräte" in der Fristbox gelöscht und vollkommen neuangelegt um wirklich sicher zu gehen das ich da nicht irgendwo einen kleinen Dreher drin habe, den ich vielleicht immer und immer wieder übersehe. Leider ohne erfolg.

Spoiler: aktuelle SIP(MAC-Adresse).cnf

# SIP Configuration Generic File

# Call Manager
call_manager1_addr: "192.168.178.1"
call_manager1_sip_port: 5060
call_manager2_addr: "192.168.178.1"
call_manager2_sip_port: 5060

# Line 1
proxy1_address: "USECALLMANAGER"
proxy1_port: 5060
line1_name: "**620"
line1_authname: "IpTelefon1"
line1_password: "Passwort1234"
line1_displayname: "Patrick L1"
line1_shortname: "Leitung1"

# Line 2
proxy2_address: "USECALLMANAGER"
proxy2_port: 5060
line2_name: "**621"
line2_authname: "IpTelefon2"
line2_password: "Passwort1234"
line2_displayname: "Patrick L2"
line2_shortname: "Leitung2"

phone_label: "Patrick "

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "Mein Telefon"

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco"

Ich habe auch mit "line1_name" probiert ob es 620 oder **620 heißen muss - egal wie, Erfolg brachte es leider nicht. Ich bin auf einigen anderen Websites noch über eine Parameter Namens "<transportLayerProtocol>2</transportLayerProtocol> <!-- 2 = UDP / Bei Verwendung an FritzBox nötig -->" gestolpert. Das hat wohl was mit der Kommunikation mit der FritzBox zu tun, neuere OS-Versionen des Routers benötigen hier wohl eine gesonderte Zuweisung des Wertes manche schreiben 1 ist richtig, andere sagen da muss 4 stehen ... ich habe alles von 1-4 probiert, leider brachte auch hier nichts Erfolg. Ich füge hier mal noch ein paar Screenshots der Einstellungen an (Telefonnummer sind "geschwärzt" worden) ...

 

[sonyKatze]

Ein anderer Ansatz wäre, damit Du nicht so blind umher konfigurierst, den Datenverkehr auf der Schnittstelle LAN mitzuschneiden. Wir helfen dann gerne beim Lesen der Mitschnitte mittels der App Wireshark.

 

[koyaanisqatsi]

Moinsen


Ich vermute mal ganz stark, dass du ( @Patrix2911 ) ganz knapp vor einer erfolgreichen Registrierung gescheitert bist, weil du es (noch) nicht ganz kapiert hast.
Es gelten strikt die "Anmeldedaten" aus der FRITZ!Box, nix Anderes, also vergiss die internen Nummern (**620 und 620).
Auch der Authorisierungsname ist/sollte unnötig sein, wenn der Benutzername genommen wird.
Also mein heiss ans Herz gelegter Tipp für eine erfolgreiche Registration für einen Account wäre demzufolge (2. Screenshot: Anmeldedaten)...

line1_name: "IpTelefon1"
line1_authname: ""
line1_password: "Kennwort aus den IpTelefon1 Anmeldedaten"

 

[Patrix2911]

Vielen Dank für eure Hilfe ... ich weiss nicht genau warum es jetzt aber nachfolgende Konfiguration klappt jetzt bei mir (lediglich die Passwörter habe ich jetzt zum Veröffentlichen nicht geändert.

Spoiler: SIPDefault.cnf

;SIPDefault.cnf

# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-8-12-00

# Proxy Server
proxy1_address: "192.168.178.1" ; Can be dotted IP or FQDN
proxy2_address: "" ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# 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, g729a)
preferred_codec: g711ulaw

# TOS bits in media stream [0-5] (Default - 5)
#tos_media: 5

# 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

# 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 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "" ; Example: ./sip_phone/

# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.53.103.104" ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EAST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 0 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format : D/M/Y

# 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-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 2 ; Default 0 (Disable sending all calls as anonymous)

# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101 ; Default 101

# Sync value of the phone used for remote reset
sync: 1 ; Default 1

####### New Parameters added in Release 2.1 #######

# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable

####### New Parameters added in Release 2.2 ######

# NAT/Firewall Traversal
nat_enable: 0 ; 0-Disabled (default), 1-Enabled
nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: "192.168.178.1" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060

####### New Parameter added in Release 3.0 #######

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)

####### New Parameters added in Release 3.1 #######

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged

####### New Parameters added in Release 4.0 #######
; 0-Disabled (default), 1-Enabled

# XML URLs
services_url: "" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: "" ; URL for branding logo to be used on phone display

# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP

# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled

####### New Parameters added in Release 4.4 #######

# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)

####### New Parameters added in Release 6.0 #######

# Dialtone Stutter for MWI
stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled

#Voice Mail extention
messages_uri: 8500

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0

#Transfer by hanging up the phone
transfer_onhook_enabled:1

Spoiler: SIP(MAC-Adresse).cnf

# SIP Configuration Generic File

# Call Manager
call_manager1_addr: "192.168.178.1"
call_manager1_sip_port: 5060
call_manager2_addr: "192.168.178.1"
call_manager2_sip_port: 5060

# Line 1
proxy1_address: "USECALLMANAGER"
proxy1_port: 5060
line1_name: "IpTelefon1"
line1_authname: "IpTelefon1"
line1_password: "Passwort1234"
line1_displayname: "Patrick L1"
line1_shortname: "Leitung1"

# Line 2
proxy2_address: "USECALLMANAGER"
proxy2_port: 5060
line2_name: "IpTelefon2"
line2_authname: "IpTelefon2"
line2_password: "Passwort1234"
line2_displayname: "Patrick L2"
line2_shortname: "Leitung2"

phone_label: "Patrick "

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "Mein Telefon"

# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco"

Ich habe übrigens mal nachgeschaut ... 10 Jahre, stand das Ding schon hier im Schrank herum.