11-18-2012 02:28 AM
Please CHeck the Below Configuration....here the Connection plar
three voice-port 0/2/0 , 1 , 2 are configured with Connection plar 1201 , 1202 , 1203 respectively.
Now i want it in reverse way. when ever a 1201 DN Phone calls to PSTN , it should select only Voice-port 0/2/0
and whenever 1202 Phone Makes a PSTN Call it should go through only voice-port 0/2/1 etc.
BR1#show run
Building configuration...
Current configuration : 2500 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname BR1
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
memory-size iomem 5
ip cef
!
!
!
!
no ip domain lookup
!
multilink bundle-name authenticated
!
!
!
trunk group outside
description OUTGOING TRUNK GROUP
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
!
!
!
!
!
!
!
!
!
voice register global
system message "9971 SIP Phone in SRST MODE"
max-dn 4
max-pool 4
!
voice register pool 1
id network 192.168.10.0 mask 255.255.255.0
!
!
!
!
!
!
!
archive
log config
hidekeys
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 192.168.10.221 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!
!
!
!
!
!
!
control-plane
!
!
!
!
!
!
!
dial-peer voice 1 voip
description TO CUCM PUB
preference 1
destination-pattern 1...
session target ipv4:192.168.10.101
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
voice-class codec 1
voice-class h323 1
!
dial-peer voice 2 voip
description To CUCM SUB
destination-pattern 1...
session target ipv4:192.168.10.102
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
voice-class codec 1
voice-class h323 1
!
dial-peer voice 3 pots
trunkgroup outside
description To HQ in WAN FAILURE
preference 2
destination-pattern 11..
prefix 769 ## COnverting to FUll length PSTN Number 76511.. of HQ ###
!
dial-peer voice 4 pots
trunkgroup outside
description To BR2 in WAN FAILURE
preference 2
destination-pattern 13..
prefix 789
!
dial-peer voice 101 pots
trunkgroup outside
description OUTGOING International CALLS
destination-pattern 900T
prefix 00
!
dial-peer voice 102 pots
trunkgroup Outside
description Outgoing National Calls
destination-pattern 90[1234678].......
forward-digits 9
!
dial-peer voice 103 pots
trunkgroup Outside
description Outgoing local calls
destination-pattern 9[1234567]......
forward-digits 7
!
!
voice-port 0/2/0 ###FXO Ports
connection-plar opx 1201
bearer-cap speech
trunk-group outside
voice-port 0/2/1
connection-plar opx 1202
bearer-cap speech
trunk-group outisde
voice-port 0/2/2
connection-plar opx 1203
bearer-cap speech
trunk-group outside
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 192.168.10.221 port 2000
max-ephones 3
max-dn 6 dual-line
system message primary "SCCP Phonesin SRST Mode"
keepalive 10
!
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
login
!
!
end
BR1#
11-19-2012 11:14 AM
mohamedbineesh,
This community does not provide technical support and is not staffed with technical support experts. I recommend you post this and future technical support questions to the Cisco Support Community (https://supportforums.cisco.com/index.jspa) where our Cisco technical support experts provide debugging assistance. Another option is to open a ticket with the Cisco Technical Assistance Center (www.cisco.com/go/support) to get expert assistance.
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