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SRST Router Outbound Call Configuration

mohamedbineesh
Level 1
Level 1

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#

1 Reply 1

keglass
Level 7
Level 7

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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