Hi all, I am searching for a way to have the communication between VOIP-Gateway (ISR4000) and IP-Phone, between VOIP-Gateway and CUCM in the global-Routing-table, and the communication between VOIP-Gateway and SIP-provider in a VRF for security-reasons. Is that supported in any way ?
Config looks like following :
interface Loopback1
description Loopback-Interface for Voice-Applications
ip address 10.224.255.5 255.255.255.252
ip pim sparse-mode
h323-gateway voip interface
!
interface GigabitEthernet0/0.751
description IP-Phones_Vlan751
encapsulation dot1Q 751
ip address 10.224.51.1 255.255.255.0
ip access-group ip-traffic-voice-in in
no ip redirects
no ip proxy-arp
ip accounting output-packets
ip pim sparse-mode
ip verify unicast source reachable-via rx
ip tcp adjust-mss 1320
service-policy input xQOS-MARK-POLICY
!
interface GigabitEthernet0/2
description SIP-Connect
!#### ip vrf forwarding FVRF-SIP-TRUNK ->>>> this Interface should be in VRF
ip address 192.168.1.254 255.255.255.0
ip access-group SIP-ACL-TOLL-Fraud in
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow monitor FLOW_MONITOR input
ip flow monitor FLOW_MONITOR output
load-interval 30
duplex auto
speed auto
no cdp enable
!
ip route 162.218.251.159 255.255.255.255 192.168.1.1
!#### ip route FVRF-SIP-TRUNK 162.218.251.159 255.255.255.255 GigabitEthernet0/2 192.168.1.1
!
ip access-list extended SIP-ACL-TOLL-Fraud
permit ip host 192.168.1.1 host 192.168.1.254
permit ip host 162.218.251.159 host 192.168.1.254
deny ip any any
!
dial-peer voice 1001 voip
description *SIP Trunk from CUCM*
session protocol sipv2
voice-class codec 7000
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1000 voip
description *SIP Trunk to CUCM*
destination-pattern T
session protocol sipv2
session server-group 1
voice-class codec 7000
no voice-class sip outbound-proxy
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3001 voip
description incoming from SIP_PROVIDER
translation-profile incoming TELE2SIPIN
service auto_att_flash
session protocol sipv2
session target ipv4:162.218.251.159:5083
incoming called-number +43723485085T
incoming uri from 1
no voice-class sip outbound-proxy
voice-class sip profiles 1
voice-class sip options-keepalive
voice-class sip copy-list 1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 3000 voip
description to SIP_PROVIDER
translation-profile outgoing TELE2SIPOUT
destination-pattern T
session protocol sipv2
session target ipv4:162.218.251.159:5083
voice-class codec 7000
no voice-class sip associate registered-number
no voice-class sip outbound-proxy
voice-class sip profiles 1
voice-class sip copy-list 1
dtmf-relay rtp-nte
no vad
!
sip-ua
credentials ....
authentication ....
retry invite 3
retry register 3
timers trying 1000
registrar ipv4:162.218.251.159:5083 expires 3600
connection-reuse
!
!
