Pagina 1 di 1
1751-V e VoipBuster
Inviato: mer 17 ott , 2007 5:56 pm
da orion
Salve ho un router 1751-V con cui imbastire un proxy SIP ed utilizzare il provider voipbuster.
Visto che sono abbastanza nuovo, vi chiedo se la versione dell'IOS che c'è installata va bene. In caso contrario quale dovrei utilizzare?
Grazie in anticipo!
Vi posto l'sh ver
Codice: Seleziona tutto
sh ver
Cisco Internetwork Operating System Software
IOS (tm) C1700 Software (C1700-SV8Y-M), Version 12.2(11)T7, RELEASE SOFTWARE (f c1)
TAC Support: http://www.cisco.com/tac
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Fri 28-Feb-03 12:27 by dchih
Image text-base: 0x80008124, data-base: 0x81136984
ROM: System Bootstrap, Version 12.2(1r)XE1, RELEASE SOFTWARE (fc1)
mercurio uptime is 2 days, 21 hours, 35 minutes
System returned to ROM by power-on
System image file is "flash:c1700-sv8y-mz.122-11.T7.bin"
cisco 1751 (MPC860P) processor (revision 0x200) with 55706K/9830K bytes of memor y.
Processor board ID JAD061205FM (2489840274), with hardware revision 0000
MPC860P processor: part number 5, mask 2
Bridging software.
X.25 software, Version 3.0.0.
1 Ethernet/IEEE 802.3 interface(s)
1 FastEthernet/IEEE 802.3 interface(s)
32K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)
Configuration register is 0x2102
Inviato: mar 13 nov , 2007 10:42 pm
da orion
Allora ho aggiornato la IOS ora ho
Codice: Seleziona tutto
Cisco IOS Software, C1700 Software (C1700-SPSERVICESK9-M), Version 12.4(9)T1
con la seguente configurazione
Codice: Seleziona tutto
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
no logging buffered
enable secret 5 XXXXXXXXXX
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
aaa session-id common
!
resource policy
!
voice-card 1
!
voice-card 2
!
ip cef
!
!
!
!
ip name-server 212.216.172.62
!
!
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
!
!
!
!
voice register global
mode cme
max-dn 144
max-pool 24
!
voice register dn 10
number 2000
!
voice register pool 1
id mac AAAA.BBBB.CCCC
number 1 dn 10
!
!
!
!
!
username XXXXXXXXXX privilege 15 secret 5 XXXXXXXXXX
!
!
!
!
!
interface ATM0/0
description INTERFACCIA ATM/ADSL
bandwidth 4382
no ip address
no ip route-cache cef
no ip route-cache
no ip mroute-cache
no atm ilmi-keepalive
dsl operating-mode auto
hold-queue 224 in
pvc 8/35
encapsulation aal5mux ppp dialer
dialer pool-member 1
!
!
interface FastEthernet0/0
description $ETH-LAN$
ip address 192.168.0.1 255.255.255.0
ip nat inside
speed auto
!
interface Dialer0
description DIALER ALICE-ADSL
bandwidth 4382
ip address negotiated
ip nat outside
encapsulation ppp
ip tcp header-compression
dialer pool 1
dialer-group 1
ppp pap sent-username XXXXXXXXXX password 7 XXXXXXXXXX
!
ip default-gateway 192.168.0.1
ip route 0.0.0.0 0.0.0.0 Dialer0
!
!
ip http server
no ip http secure-server
ip nat inside source list 101 interface Dialer0 overload
ip nat inside source static tcp 192.168.0.2 4949 interface Dialer0 4949
ip nat inside source static tcp 192.168.0.2 5060 interface Dialer0 5060
ip nat inside source static tcp 192.168.0.2 4711 interface Dialer0 99
ip nat inside source static tcp 192.168.0.2 8080 interface Dialer0 8080
ip nat inside source static tcp 192.168.0.2 80 interface Dialer0 80
ip nat inside source static tcp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static tcp 192.168.0.2 22 interface Dialer0 10022
ip nat inside source static tcp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static udp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static udp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static tcp 192.168.0.2 23 interface Dialer0 23
ip nat inside source static tcp 192.168.0.2 9090 interface Dialer0 9090
!
access-list 101 permit ip host 192.168.0.2 any
!
control-plane
!
!
!
voice-port 1/0
!
voice-port 1/1
!
!
!
!
!
!
dial-peer voice 1 voip
session target ipv4:192.168.1.8
incoming called-number 18
codec g711ulaw
!
sip-ua
authentication username XXXXXXXXXX password XXXXXXXXXX
retry invite 4
retry response 3
retry bye 2
retry cancel 2
registrar ipv4:194.221.62.198:5060 expires 3600
sip-server ipv4:194.221.62.198:5060
!
!
telephony-service
max-conferences 4 gain -6
transfer-system full-consult
!
!
line con 0
line aux 0
line vty 0 4
password 7 XXXXXXXXXX
!
end
Ma devo aver fatto confusione,
perchè non si registra ovvero ho l'impressione che non ci provi proprio,
infatti se tento con
ottengo il seguente risultato
Codice: Seleziona tutto
Line peer expires(sec) registered
============ ============= ============ ===========
qualche dritta??
grazie in anticipo!!
Inviato: mar 20 nov , 2007 10:10 pm
da orion
Salve,
ho fatto alcune modifiche alla configurazione precedente ed ora riesco far chiamare gli interni fra loro (int 10 12 e 14). Ora dovrei poter chiamare all'esterno tramite il provider sip esterno, ma quando faccio un numero ricevo il messaggio di errore "not implemented".
suppongo l'errore sia nel modo in cui ho gestito il dial plan o il dial pattern.
vi posto la conf.
Vi ringrazio anticipatamente
Codice: Seleziona tutto
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
no logging buffered
enable secret 5 XXXXXXXXXXXXXXXXXXXXXX
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
aaa session-id common
!
resource policy
!
clock timezone GTM1 1
voice-card 1
!
voice-card 2
!
ip cef
!
!
!
!
ip name-server 212.216.172.62
!
!
voice call send-alert
voice call disc-pi-off
voice call debug short-header
voice rtp send-recv
voice dsp release early
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
!
!
!
!
!
!
!
voice register global
mode cme
max-dn 144
max-pool 24
!
voice register dn 10
number 10
name 10
label 10
!
voice register dn 12
number 12
name 12
label 12
!
voice register dn 14
number 14
name 14
label 14
!
voice register pool 1
id mac AAAA.AAAA.1BFE
number 2 dn 14
codec g711alaw
!
voice register pool 2
id mac BBBB.BBBB.CB96
number 2 dn 12
codec g711alaw
!
voice register pool 3
id mac CCCC.CCCC.80A4
number 1 dn 10
codec g711alaw
!
!
!
!
!
username root privilege 15 secret 5 XXXXXXXXXXXXXXXXXXXXXX
!
!
!
!
!
interface ATM0/0
description INTERFACCIA ATM/ADSL
bandwidth 4382
no ip address
no ip route-cache cef
no ip route-cache
no ip mroute-cache
no atm ilmi-keepalive
dsl operating-mode auto
hold-queue 224 in
pvc 8/35
encapsulation aal5mux ppp dialer
dialer pool-member 1
!
!
interface FastEthernet0/0
description LAN LOCALE
ip address 192.168.0.1 255.255.255.0
ip nat inside
speed auto
!
interface Dialer0
description DIALER ALICE-ADSL
bandwidth 4382
ip address negotiated
ip nat outside
encapsulation ppp
ip tcp header-compression
dialer pool 1
dialer-group 1
ppp pap sent-username XXXXXXXXXXXXXXXXXXXXXX password 7 XXXXXXXXXXXXXXXXXXXXXX
!
ip default-gateway 192.168.0.1
ip route 0.0.0.0 0.0.0.0 Dialer0
!
!
ip http server
no ip http secure-server
ip nat inside source list 101 interface Dialer0 overload
ip nat inside source static tcp 192.168.0.2 4949 interface Dialer0 4949
ip nat inside source static tcp 192.168.0.2 4711 interface Dialer0 99
ip nat inside source static tcp 192.168.0.2 8080 interface Dialer0 8080
ip nat inside source static tcp 192.168.0.2 80 interface Dialer0 80
ip nat inside source static tcp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static tcp 192.168.0.2 22 interface Dialer0 10022
ip nat inside source static tcp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static udp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static udp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static tcp 192.168.0.2 23 interface Dialer0 23
ip nat inside source static tcp 192.168.0.2 9090 interface Dialer0 9090
!
access-list 101 permit ip 192.168.0.0 0.0.0.255 any
!
control-plane
!
!
!
voice-port 1/0
!
voice-port 1/1
!
!
!
!
!
!
dial-peer voice 1 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:194.221.62.198
!
!
sip-ua
authentication username XXXXXXXXXXXXXXXXXXXXXX password XXXXXXXXXXXXXXXXXXXXXX
retry invite 4
retry response 3
retry bye 2
retry cancel 2
registrar ipv4:194.221.62.198:5060 expires 3600
sip-server ipv4:194.221.62.198:5060
!
!
telephony-service
max-conferences 4 gain -6
transfer-system full-consult
!
!
line con 0
line aux 0
line vty 0 4
password 7 XXXXXXXXXXXXXXXXXXXXXX
!
ntp clock-period 17208428
ntp source Dialer0
ntp server 193.204.114.232
end
EUREKA!
Inviato: dom 25 nov , 2007 1:05 am
da orion
Finalmente funziona!
Ovvero
gli interni si chiamano fra loro e riescono a chiamare fuori
vi posto la configurazione funzionante!
Codice: Seleziona tutto
Current configuration : 4178 bytes
!
! Last configuration change at 18:02:30 GTM1 Sat Nov 24 2007 by root
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname minosse
!
boot-start-marker
boot-end-marker
!
no logging buffered
enable secret 5 XXXXXXXXXXXXXXXX
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
aaa session-id common
!
resource policy
!
clock timezone GTM1 1
voice-card 1
!
voice-card 2
!
ip cef
!
!
!
!
ip domain name xxxxxxxxx.xxx
ip name-server 212.216.172.62
ip ssh authentication-retries 2
ip ssh version 2
!
!
voice call send-alert
voice call disc-pi-off
voice call debug short-header
voice rtp send-recv
voice dsp release early
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
!
!
!
!
!
!
!
voice register global
mode cme
max-dn 144
max-pool 24
!
voice register dn 10
number 10
name 10
label 10
!
voice register dn 12
number 12
name 12
label 12
!
voice register dn 14
number 14
name 14
label 14
!
voice register dn 16
number 16
name 16
label 16
!
voice register pool 1
id mac 0000.0000.0000
number 2 dn 14
codec g711alaw
!
voice register pool 2
id mac 0000.0000.0000
number 2 dn 12
codec g711alaw
!
voice register pool 3
id mac 0000.0000.0000
number 1 dn 16
codec g711alaw
!
voice register pool 4
id mac 0000.0000.0000
number 1 dn 16
username 16 password 16
codec g711alaw
!
!
!
!
!
username XXXXXXXXXXXXXXXX privilege 15 secret 5 XXXXXXXXXXXXXXXX
!
!
!
!
!
interface ATM0/0
description INTERFACCIA ATM/ADSL
bandwidth 4382
no ip address
no ip route-cache cef
no ip route-cache
no ip mroute-cache
no atm ilmi-keepalive
dsl operating-mode auto
hold-queue 224 in
pvc 8/35
encapsulation aal5mux ppp dialer
dialer pool-member 1
!
!
interface FastEthernet0/0
description LAN LOCALE
ip address 192.168.0.1 255.255.255.0
ip nat inside
speed auto
!
interface Dialer0
description DIALER ALICE-ADSL
bandwidth 4382
ip address negotiated
ip nat outside
encapsulation ppp
ip tcp header-compression
dialer pool 1
dialer-group 1
ppp pap sent-username XXXXXXXXXXXXXXXX password 7 XXXXXXXXXXXXXXXX
!
ip default-gateway 192.168.0.1
ip route 0.0.0.0 0.0.0.0 Dialer0
!
!
ip http server
no ip http secure-server
ip nat inside source list 101 interface Dialer0 overload
ip nat inside source static tcp 192.168.0.2 9090 interface Dialer0 9090
ip nat inside source static tcp 192.168.0.2 23 interface Dialer0 23
ip nat inside source static udp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static udp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static tcp 192.168.0.2 4242 interface Dialer0 4242
ip nat inside source static tcp 192.168.0.2 22 interface Dialer0 10022
ip nat inside source static tcp 192.168.0.2 4662 interface Dialer0 4662
ip nat inside source static tcp 192.168.0.2 80 interface Dialer0 80
ip nat inside source static tcp 192.168.0.2 8080 interface Dialer0 8080
ip nat inside source static tcp 192.168.0.2 4711 interface Dialer0 99
ip nat inside source static tcp 192.168.0.2 4949 interface Dialer0 4949
!
access-list 101 permit ip 192.168.0.0 0.0.0.255 any
!
control-plane
!
!
!
voice-port 1/0
!
voice-port 1/1
!
!
!
!
!
!
dial-peer voice 1 voip
preference 1
destination-pattern 3T
session protocol sipv2
session target ipv4:194.221.62.198
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 2 voip
preference 1
destination-pattern 0T
session protocol sipv2
session target ipv4:194.221.62.198
dtmf-relay rtp-nte
codec g711alaw
no vad
!
sip-ua
authentication username XXXXXXXXXXXXXXXX password XXXXXXXXXXXXXXXX
retry invite 4
retry response 3
retry bye 2
retry cancel 2
registrar ipv4:194.221.62.198:5060 expires 3600
sip-server ipv4:194.221.62.198:5060
!
!
telephony-service
max-conferences 4 gain -6
transfer-system full-consult
!
!
line con 0
line aux 0
line vty 0 4
password 7 XXXXXXXXXXXXXXXX
transport input ssh
!
ntp clock-period 17208399
ntp source Dialer0
ntp server 193.204.114.232
end
A questo punto mi piacerebbe sapere se è possibile impostare il router affinchè invii, nelle chiamate SIP, il caller ID associato al mio account VB.
Ho già verificato il mio numero di telefono, da client saltuariamente il tutto funziona (se chiamo dal client sul cellulare appare l'id chiamante).
Qualcuno ha qualche dritta??????????????????
grazie in anticipo.