Ora io riesco a far uscire gli analogici, ma quelli registrati sul call manager non escono proprio.
Lascio la mia config, e un debug che mostra cosa succede quando chiamo da CUCM 7.
Codice: Seleziona tutto
Building configuration...
Current configuration : 2600 bytes
!
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource policy
!
!
!
ip name-server 151.99.0.100
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
ip circuit max-calls 1000
ip circuit carrier-id AA reserved-calls 200
call start interwork
sip
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 2
call start fast
!
!
!
!
!
!
!
voip-incoming translation-profile GATEWAY
!
!
translation-rule 1
Rule 1 ^0 00390
!
!
translation-rule 2
Rule 0 ^1001 ******
!
!
!
!
interface Ethernet0
ip address 192.168.1.70 255.255.255.0
hold-queue 100 out
!
interface ATM0
no ip address
shutdown
no atm ilmi-keepalive
dsl operating-mode auto
!
ip route 0.0.0.0 0.0.0.0 192.168.1.100
ip http server
!
!
!
control-plane
!
!
voice-port 1
ring cadence pattern01
echo-cancel coverage 32
no vad
cptone IT
timeouts interdigit 5
bearer-cap Speech
caller-id enable
!
voice-port 2
ring cadence pattern01
cptone IT
!
voice-port 3
ring cadence pattern01
cptone IT
!
voice-port 4
ring cadence pattern01
cptone IT
!
dial-peer voice 1 pots
destination-pattern ******
port 1
!
dial-peer voice 10 voip
shutdown
destination-pattern .T
translate-outgoing calling 2
translate-outgoing called 1
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte digit-drop h245-alphanumeric
fax rate 9600
fax protocol pass-through g711alaw
!
dial-peer voice 20 voip
destination-pattern 10.T
voice-class h323 1
session target ipv4:192.168.1.20
!
dial-peer voice 30 voip
destination-pattern .T
translate-outgoing calling 2
translate-outgoing called 1
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte digit-drop h245-alphanumeric
fax rate 9600
fax protocol pass-through g711alaw
!
gateway
emulate cisco h323 bandwidth
!
sip-ua
authentication username ******* password ******* retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 10
registrar dns:sip.skype.com:5060 expires 3600
sip-server dns:sip.skype.com:5060
no suspend-resume
!
!
line con 0
line vty 0 4
login
!
scheduler max-task-time 5000
end
IL DEBUG
Codice: Seleziona tutto
The Call Setup Information is:
Call Control Block (CCB) : 0x8242DEC0
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : ************
Called Number : 003901******
Source IP Address (Sig ): 192.168.1.70
Destn SIP Req Addr:Port : 0.0.0.0:5060
Destn SIP Resp Addr:Port : 0.0.0.0:5060
Destination Name : sip.skype.com
*Mar 8 17:03:56.070: //311/807959F10200/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 200