Kein Anruf möglich cp-7960 und cisco 1750

[ShiningStar]

Ich hab oben genannte Hardware. Ich hab auch die DNS von t-online ins telefon eingetragen, und dann stand oben plötzlich korrektes datum und Uhrzeit. Soweit ok. Dann wollte ich das in EU format ändern, da war es plötzlich weg. Viel schlimmer jedoch ist, das ich nun weder angerufen werden kann, noch anrufen kann. Meine konfig sieht wie folgt aus:

DialIn_LTB_01#sh conf
Using 2726 out of 29688 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname DialIn_LTB_01
!
enable secret 5 xxx
enable password xxx
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
vpdn enable
!
vpdn-group 1
request-dialin
protocol pppoe
ip mtu adjust
!
!
!
!
!
!
interface Ethernet0
description T-DSL Hello World
no ip address
ip access-group 100 in
ip mtu 1492
ip nat outside
no ip mroute-cache
half-duplex
pppoe enable
pppoe-client dial-pool-number 21
no cdp enable
!
interface FastEthernet0
description inside lan
ip address 192.168.0.2 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat inside
ip tcp adjust-mss 1452
no ip mroute-cache
speed auto
full-duplex
no cdp enable
!
interface Dialer21
ip address negotiated
ip access-group 100 in
ip mtu 1492
ip nat outside
encapsulation ppp
ip tcp adjust-mss 1452
no ip mroute-cache
dialer pool 21
dialer-group 2
no cdp enable
ppp authentication pap callin
ppp pap sent-username [email protected] password 0 xxx
ppp ipcp dns request
!
ip nat inside source list 101 interface Dialer21 overload
ip nat inside source static tcp 192.168.0.5 5060 interface Dialer1 5060
ip classless
ip route 0.0.0.0 0.0.0.0 Dialer21
no ip http server
!
!
access-list 100 permit icmp any any echo-reply
access-list 100 permit icmp any any packet-too-big
access-list 100 permit icmp any any echo
access-list 100 permit icmp any any ttl-exceeded
access-list 100 permit tcp any any established
access-list 100 permit udp any eq domain any
access-list 100 permit udp any eq ntp any
access-list 101 permit udp any any eq ntp
access-list 101 permit udp any any eq domain
access-list 101 permit tcp any any eq smtp
access-list 101 permit tcp any any eq pop3
access-list 101 permit tcp any any eq 143
access-list 101 permit tcp any any eq www
access-list 101 permit tcp any any eq 443
access-list 102 deny udp any eq netbios-dgm any
access-list 102 deny udp any eq netbios-ns any
access-list 102 deny udp any eq netbios-ss any
access-list 102 deny udp any range snmp snmptrap any
access-list 102 deny udp any range bootps bootpc any
access-list 102 deny tcp any eq 137 any
access-list 102 deny tcp any eq 138 any
access-list 102 deny tcp any eq 139 any
access-list 102 permit ip any any
dialer-list 2 protocol ip list 101
no cdp run
!
snmp-server community LTB1 RW
snmp-server enable traps tty
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
exec-timeout 0 0
no login
!
ntp server 192.53.103.104
ntp server 129.69.1.153
ntp server 130.149.17.8
ntp server 131.188.3.220
end

Ich weiss echt nicht mehr weiter, helft mir mal bitte. Danke schon eimal im vorraus.

 

[hedgeschredder]

Hi,

Du musst natürlich noch die Ports für den RTP stream (analog zu dem SIP 5060) weiterleiten. In der config des phones ist auch irgendwo die port range zu finden:

start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 16387 ; End RTP range for media (default - 32766)


In der access-list 100 ist dann noch die entsprechenden Ports freizuschalten.

Cisco SIP phones verwenden UDP port 5060, also in der Portweiterleitung das Protokoll von tcp auf udp ändern.

In der config des Phones:
voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060)

ip nat inside source static udp 192.168.0.5 5060 interface Dialer21 5060

Für das inteface ethernet 0 reicht die folgende config:

interface Ethernet0/0
no ip address
no ip mroute-cache
half-duplex
pppoe enable
pppoe-client dial-pool-number 21
no cdp enable

Probier das dochmal aus, später kann man noch ppp multilink implementieren um bei hoher Auslastung noch problemlos telefonieren zu können.

Ich schlage vor das diese config dir Grundlage für alles weitere ist.

 

[hedgeschredder]

P.S.

kommst Du mit dem Teil ins Internet ? Folgendes müsste noch geändert werden:

access-list 1 permit 192.168.0.0 0.0.0.255
ip nat inside source list 1 interface Dialer21 overload
dialer-list 2 protocol ip permit

 

[ShiningStar]

sicherlich komme ich mit dem teil ins inet, sonst könnte ich hier nicht schreiben, warum fragst du ?

 

[ShiningStar]

Wie meinst du das mit auf udp ändern und wie muss ich denn die konfig machen um ein port range frei zu geben und weiter zu leiten?

 

[hedgeschredder]

Hi,

ip nat inside source static udp 192.168.0.5 5060 interface Dialer21 5060

Port ranges gehen nicht, aber man kann die 4 ports analog eintragen:

ip nat inside source static udp 192.168.0.5 16384 interface Dialer21 16384
ip nat inside source static udp 192.168.0.5 16385 interface Dialer21 16385
ip nat inside source static udp 192.168.0.5 16386 interface Dialer21 16386
ip nat inside source static udp 192.168.0.5 16387 interface Dialer21 16387

 

[ShiningStar]

also meine config sieht nun wie folgt aus. Ins i net komme ich, telefonieren oder angerufen werden geht noch immer nicht:

sh conf
Using 2598 out of 29688 bytes
!
! Last configuration change at 22:44:20 UTC Thu Aug 11 2005
! NVRAM config last updated at 22:44:22 UTC Thu Aug 11 2005
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
enable secret 5 xxx
enable password xxx
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
vpdn enable
!
vpdn-group 1
request-dialin
protocol pppoe
ip mtu adjust
!
!
!
!
!
!
interface Ethernet0
description T-DSL Hello world
no ip address
ip access-group 100 in
ip mtu 1492
ip nat outside
no ip mroute-cache
half-duplex
pppoe enable
pppoe-client dial-pool-number 21
no cdp enable
!
interface FastEthernet0
description inside Lan
ip address 192.168.0.1 255.255.255.0
no ip unreachables
no ip proxy-arp
ip nat inside
ip tcp adjust-mss 1452
no ip mroute-cache
speed auto
full-duplex
no cdp enable
!
interface Dialer21
description Dialerinterface dsl
ip address negotiated
ip access-group 100 in
ip mtu 1492
ip nat outside
encapsulation ppp
ip tcp adjust-mss 1452
no ip mroute-cache
dialer pool 21
dialer-group 2
no cdp enable
ppp authentication pap callin
ppp pap sent-username xxx#[email protected] password 0 xxx
ppp ipcp dns request
!
ip nat inside source list 101 interface Dialer21 overload
ip nat inside source static tcp 192.168.0.5 5060 interface Dialer21 5060
ip classless
ip route 0.0.0.0 0.0.0.0 Dialer21
ip http server
!
!
access-list 100 permit icmp any any echo-reply
access-list 100 permit icmp any any packet-too-big
access-list 100 permit icmp any any echo
access-list 100 permit icmp any any ttl-exceeded
access-list 100 permit tcp any any established
access-list 100 permit udp any eq domain any
access-list 100 permit udp any eq ntp any
access-list 101 permit udp any any eq ntp
access-list 101 permit udp any any eq domain
access-list 101 permit tcp any any eq smtp
access-list 101 permit tcp any any eq pop3
access-list 101 permit tcp any any eq 143
access-list 101 permit tcp any any eq www
access-list 101 permit tcp any any eq 443
access-list 101 permit udp any any eq 5060
access-list 101 permit udp any any range 16384 32766
access-list 101 permit tcp any any range 16384 32766
dialer-list 2 protocol ip list 101
no cdp run
!
snmp-server community xxx
snmp-server enable traps tty
!
line con 0
line aux 0
line vty 0 4
password xxx
login
!
ntp clock-period 17179790
ntp server 192.53.103.104
ntp server 129.69.1.153
ntp server 130.149.17.8
ntp server 131.188.3.220
end

Ich weiss nich wie ich den rtp stream weiter leiten soll, oder was schief geht... irgendwie geht das mit dem telefon nicht. Helft mir bitte

 

[ShiningStar]

problem ist auch das mein telefon immer nur die standart media ports nimmt und ich daran leider nichts ändern kann... ich kann doch keine 1000 nat einträge nur für das tele machen... das is n höllen act....

 

[ShiningStar]

sodele, config update gemacht... siehe da, nun kann ich zwar anrufen, er verbindet, aber hören tue ich nichts, auch zeigt das telefon mir kein datum und uhrzeit mehr an und angerufen werden kann ich auch nicht. Mein router hat folgende config:

sh conf
Using 3269 out of 29688 bytes
!
! Last configuration change at 03:03:43 UTC Fri Aug 12 2005
! NVRAM config last updated at 03:03:45 UTC Fri Aug 12 2005
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
enable secret 5 xxx
enable password xxx
!
ip subnet-zero
!
!
!
ip audit notify log
ip audit po max-events 100
vpdn enable
!
vpdn-group 1
request-dialin
protocol pppoe
ip mtu adjust
!
!
!
!
!
!
interface Ethernet0
description T-DSL Hello world
no ip address
ip access-group 100 in
ip mtu 1492
ip nat outside
no ip mroute-cache
half-duplex
pppoe enable
pppoe-client dial-pool-number 21
no cdp enable
!
interface FastEthernet0
description inside Lan
ip address 192.168.0.1 255.255.255.0
no ip unreachables
no ip proxy-arp
ip nat inside
ip tcp adjust-mss 1452
no ip mroute-cache
speed auto
full-duplex
no cdp enable
!
interface Dialer21
description Dialerinterface dsl
ip address negotiated
ip access-group 100 in
ip mtu 1492
ip nat outside
encapsulation ppp
ip tcp adjust-mss 1452
no ip mroute-cache
dialer pool 21
dialer-group 2
no cdp enable
ppp authentication pap callin
ppp pap sent-username xxx#[email protected] password 0 xxx
ppp ipcp dns request
!
ip nat inside source list 101 interface Dialer21 overload
ip nat inside source static udp 192.168.0.5 5005 interface Dialer21 5005
ip nat inside source static udp 192.168.0.5 5004 interface Dialer21 5004
ip nat inside source static udp 192.168.0.5 5060 interface Dialer21 5060
ip nat inside source static udp 192.168.0.5 16387 interface Dialer21 16387
ip nat inside source static udp 192.168.0.5 16386 interface Dialer21 16386
ip nat inside source static udp 192.168.0.5 16385 interface Dialer21 16385
ip nat inside source static udp 192.168.0.5 16384 interface Dialer21 16384
ip classless
ip route 0.0.0.0 0.0.0.0 Dialer21
ip http server
!
!
access-list 100 permit icmp any any echo-reply
access-list 100 permit icmp any any packet-too-big
access-list 100 permit icmp any any echo
access-list 100 permit icmp any any ttl-exceeded
access-list 100 permit tcp any any established
access-list 100 permit udp any eq domain any
access-list 100 permit udp any eq ntp any
access-list 100 permit udp any any range 16384 32766
access-list 100 permit udp any any range 16384 16387
access-list 100 permit udp any any eq 5004
access-list 100 permit udp any any eq 5005
access-list 100 permit udp any any eq 5060
access-list 100 permit udp any eq 5060 any
access-list 101 permit udp any any eq ntp
access-list 101 permit udp any any eq domain
access-list 101 permit tcp any any eq smtp
access-list 101 permit tcp any any eq pop3
access-list 101 permit tcp any any eq 143
access-list 101 permit tcp any any eq www
access-list 101 permit tcp any any eq 443
access-list 101 permit udp any any eq 5060
access-list 101 permit udp any any range 16384 16387
dialer-list 2 protocol ip list 101
no cdp run
!
snmp-server community Router
snmp-server enable traps tty
!
line con 0
line aux 0
line vty 0 4
password xxx
login
!
ntp clock-period 17179785
ntp server 192.53.103.104
ntp server 129.69.1.153
ntp server 130.149.17.8
ntp server 131.188.3.220
end


Die config meines cp-7960, sipstandart.cnf:

# SIP Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
image_version: P0S3-07-4-00

# Proxy Server
proxy1_address: "sipgate.de" ; Can be dotted IP or FQDN
proxy2_address: "sipsnip.com" ; Can be dotted IP or FQDN
proxy3_address: "sipgate.de" ; 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: 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

####### 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: "ntp.sipgate.net" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; 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)
date_format: D-M-YY ; Dateformat Day, month, year

# 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
####### 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: 1 ; 0-Disabled (default), 1-Enabled
nat_address: "192.168.0.1" ; 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: 16387 ; End RTP range for media (default - 32766)
nat_received_processing: 1 ; 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
####### 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: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged

####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "http://www.fo-pa.de/cgi-bin/rss2cisco.pl" ; URL for external Phone Services
directory_url: "//niels/tftp-root/directory.xml" ; 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 (Message Waiting Indicator)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
stutter_msg_waiting: 1 ; 0-Disabled (default), 1-Enabled

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

# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
#**** 0-abgeschaltet
#**** 1-eingeschaltet


Ein problem ist es auch, das start media und end media port nicht so wie in der config auch im telefon stehen, weil der partut nicht belehren lassen will, doch meine vorgaben zu verwenden, lieber benutzt er da seine standartports... ich hoffe nicht, das ich 1600 einzelne ports freischalten muss, das mein telefon geht.

Helft mir bitte, so langsam bin ich am verzweifeln....