translate called no

[SJ SJ]

i have a scenario where incoming calls through PSTN need to be translated to a UK toll free number and going out thorough a SIP trunk provider.

Please let me know if the below mentioned config on Voice GW is Ok for this purpose.

voice call send-alert

voice call carrier capacity active

voice rtp send-recv

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

!

!

voice class codec 10

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

!

voice translation-rule 3

rule 1 /.*/ /12345/           ! ! !  12345 = UK number

!

!

voice translation-profile ITSP_Outgoing

translate called 1

!

!

!

voice-port 0/0/0

!

voice-port 0/0/1

!

voice-port 0/0/2

!

voice-port 0/0/3

!

!

dial-peer voice 111 pots

incoming called-number .T

direct-inward-dial

port 0/0/0

!

!

dial-peer voice 1 voip

translation-profile outgoing ITSP_Outgoing

destination-pattern .T

voice-class codec 1

session protocol sipv2

session target ipv4:1.1.1.1

dtmf-relay rtp-nte

no vad

!

!

sip-ua

keepalive target ipv4:1.1.1.1

authentication username 406589 password 7 06001B244F465A4854

no remote-party-id

retry invite 10

retry register 10

retry options 10

timers connect 100

registrar ipv4:1.1.1.1 expires 60

sip-server ipv4:1.1.1.1

Regards,

SJSJ

[Chris Deren]

It would be better to translate the called number on the incoming call leg instead, also this is assuming you are sending all calls to the same number.

Chris

[SJ SJ]

Thanks Chris. Is the below config ok.

voice call send-alert

voice call carrier capacity active

voice rtp send-recv

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to sip

allow-connections sip to h323

!

sip

rel1xx disable

min-se  360

header-passing

!

!

voice class codec 10

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

!

voice translation-rule 1

rule 1 /.*/ /12345/           ! ! !  12345 = UK number

!

!

voice translation-profile ITSP_Outgoing

translate called 1

!

!

!

voice-port 0/0/0

!

voice-port 0/0/1

!

voice-port 0/0/2

!

voice-port 0/0/3

!

!

dial-peer voice 111 pots

translation-profile incoming ITSP_Outgoing

incoming called-number .T

direct-inward-dial

port 0/0/0

!

dial-peer voice 112 pots

translation-profile incoming ITSP_Outgoing

incoming called-number .T

direct-inward-dial

port 0/0/1

!

!

dial-peer voice 113 pots

translation-profile incoming ITSP_Outgoing

incoming called-number .T

direct-inward-dial

port 0/0/2

!

!

dial-peer voice 114 pots

translation-profile incoming ITSP_Outgoing

incoming called-number .T

direct-inward-dial

port 0/0/3

!

dial-peer voice 1 voip

destination-pattern .T

voice-class codec 1

session protocol sipv2

session target ipv4:1.1.1.1    !!!!!!               ITSP IP

dtmf-relay rtp-nte sip-notify h245-alphanumeric

no vad

!

Regards,

SJSJ