Habe mir aus Ebay ein Cisco 7960 ersteigert mit SIP Software drauf. Möchte dieses nun mit meinen VOIP Provider kombinieren.
Funktionieren tut das ganze bereits auf einem Nortel 1140e.
Konkret geht es hier um den Provider A1 welcher folgende Konfiguration vorgibt.
User: Ihr A1.net Benutzername (Vorsicht: GroĂź-/Kleinschreibung ist wichtig! Bitte geben Sie den Namen auch OHNE den Zusatz "@A1.net" ein.)
Passwort: Das Passwort Ihres A1.net Benutzers
Domain: a1.net
Outbound-Proxy: sip.a1.net
SIP Port: 5060
Firewallfreischaltungen:
* 80.75.55.109 Port 5060 udp und tcp
* 80.75.55.93 Port 50000 - 50500 udp
* 80.75.44.240/26 Port 5060 udp und tcp, und die ports 50000 - 50500 udp
Meine Config schaut wie folgt aus:
# SIP Configuration Generic File
# Line 1
proxy1_address: "80.75.55.109"
proxy1_port: 5060
line1_name: "xxxx"
line1_authname: "xxxx"
line1_password: "xxxx"
line1_displayname: "[email protected]"
line1_shortname: "A1overIP"
####### New Parameters added in Release 2.0 #######
# All user_parameters have been removed
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: ""
# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: ""
####### New Parameters added in Release 3.0 ######
# 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: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
# Image Version
image_version: "P0S3-08-12-00"
# Emergency Proxy info
proxy_emergency: "80.75.55.109"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "80.75.55.109"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "sip.a1.net"
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "1"
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: "g711alaw"
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
# 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 this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged
phone_password: "cisco"
phone_prompt: "Cisco SIP"
# wir erlauben im Default file keine priviligierten kommandos, weil wir hier auch kein individuelles phone-passwort setzen
# 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 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "vm"
# Datumsformat
date_format: "D/M/Y"
# 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-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"
####### New Parameters added in Release 4.0 #######
# URL for external Directory location
logo_url: "http://voip.funkfeuer.at/cisco/funkfeuer.bmp" ; URL for branding logo to be used on phone display
# put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own
# 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: "193.238.157.10" ; restricted to dotted IP
# The dynamic tftp server should be set to whatever your TFTP server is. This way, it
# keeps the tftp server setting even though you might be using DHCP (default behavior
# is to use the DHCP server as a tftp server, which is rarely correct.)
# Remote Party ID
#remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
#spielt einen ton, wenn eine MWI (mailbox) nachricht eintrifft
stutter_msg_waiting: 1 ; 0 off(default, 1: on
# statistik ueber die qualitaet der leitung (default: 0/off)
# RTP-RxStat: Dur=6,Pkt=280,Oct=44800,LatePkt=0,TxLost=0,AvgJtr= 0
# RTP-TxStat: Dur=6,Pkt=272,Oct=43520
call_stats: 1 ; (0-off (default), 1-on)
# benutzet transferiert gespraech durch auflegen (0: default, user muss "transfer" knopf drucken)
#transfer_onhook_enabled : 0
#call_hold_ringback : 0
#mwi_status : 1
#sync : "1"
#xml_card_dir : ""
#xml_card_file : "CARD.XML"
# englisch ist die derzeit einzige unterstuetzte sprache
#language: english
Das Phone zeigt auch bei dem kleinen Telefonzeichen ein X an.
Wo könnte mein fehler liegen?
Vielen Dank
