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
endIL 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

