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configuring translation Rules and dial-peer

Hi

need help with dial-peer and translation patterns

i have sip line connecting to VG2921 at branch connected to CUCM over HQ VG 2811.

incoming calls are working fine with number 8062300 which will land on ext 2300

i want all incoming calls should go to 3080 which is Auto Attendent.

i tried but not able to configure that please can anyone help me with that

voice translation-rule 1

rule 1 /^5\(........\)/ /05\1/

rule 2 /^1\(.......\)/ /01\1/

rule 3 /^2\(.......\)/ /02\1/

rule 4 /^3\(.......\)/ /\1/

!

voice translation-rule 2

rule 1 /8062300/ /3080/

!

voice translation-rule 3

rule 1 /^[2-3].../ /8062300/

!

voice translation-rule 4

rule 1 /^03\(8062300\)/ /3080/

!

voice translation-rule 5

rule 1 /^77\(.......\)/ /\1/

rule 2 /^77\(.........\)/ /\1/

rule 3 /^77\(............\)/ /\1/

rule 4 /^77\(..........\)/ /\1/

!

voice translation-rule 6

rule 1 /^238\(.\)/ /03806\1/

!

!

voice translation-profile INCOM_SIP

translate calling 1

translate called 2

!

voice translation-profile INCO_SIP

translate calling 1

translate called 4

!

voice translation-profile OUT-SIP

translate calling 3

translate called 1

!

dial-peer voice 15 voip

description incoming From ITSP Server to CUCM

translation-profile incoming INCOM_SIP

destination-pattern ^80623..$

progress_ind progress enable 8

session target ipv4:192.168.12.189

voice-class codec 1

dtmf-relay rtp-nte h245-signal h245-alphanumeric

no vad

!

dial-peer voice 10 voip

translation-profile incoming INCO_SIP

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

incoming called-number ^80623..$

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711alaw

no vad

!

dial-peer voice 40 voip

translation-profile incoming INCOM_SIP

destination-pattern ^0380623..$

session target ipv4:192.168.12.189

voice-class codec 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 30 voip

description To CallManager - SBWPMPUB

destination-pattern 3080

session target ipv4:192.168.12.190

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 31 voip

description to Callmanager-subscriber

preference 1

session target ipv4:192.168.12.189

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 20 voip

description ***TO-LOCAL***

translation-profile outgoing OUT-SIP

destination-pattern .T

progress_ind progress enable 8

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.205.20.50:5060

session transport udp

voice-class codec 1

dtmf-relay rtp-nte digit-drop

no vad

!

dial-peer voice 305 voip

translation-profile incoming INCOM_SIP

rtp payload-type nte 98

session protocol sipv2

session target sip-server

session transport udp

incoming called-number 8062300

voice-class codec 1

dtmf-relay rtp-nte

no vad

5 Replies 5

Chris Deren
Hall of Fame
Hall of Fame

So, should every single call go to 3080 or all calls except for 2300?

If all calls your TR should be very simple as:

voice translation-rule 2

rule 1 /.*/ /3080/

voice translation-profile INCOM_SIP

translate calling 1

translate called 2

then apply it to the incoming dial-peer from SIP provider.

Chris

thankyou Mr.Chris,

Tried the commands but still the call ringing at ext 2300 not going to 3080

Anas Abueideh
Level 9
Level 9

Hi,

you need to configure

voice translation-rule 2

rule 1 // /3080/

if it is not working, use debug dialpeer all to check your matched dialpeers

HTH

Anas

please rate if it is helpful

Thankyou Mr.Anas

The debug voip dial peer all doc attached

dial peer 15 and 20 both are matching and while 20 is for outgoing.

Hi Mohammed,

you have issues with overlapping between the dial-peers. first you need to differentiate between incoming and outgoing dial-peers.

please apply the below config

voice service voip

  allow h323 to h323

  allow sip to sip

  allow sip to h323

  allow h323 to sip

voice translation-rule 1

rule 1 /^5\(........\)/ /05\1/

rule 2 /^1\(.......\)/ /01\1/

rule 3 /^2\(.......\)/ /02\1/

rule 4 /^3\(.......\)/ /\1/

!

voice translation-rule 2

rule 1 // /3080/

!

voice translation-rule 3

rule 1 /^[2-3].../ /8062300/

!

dial-peer voice 20 voip

description ***Incoming called-number***

translation-profile incoming INCOM_SIP

incoming called-number ^80623..$

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.205.20.50:5060

session transport udp

voice-class codec 1

dtmf-relay rtp-nte digit-drop

no vad

dial-peer voice 21 voip

description ***TO-LOCAL***

translation-profile outgoing OUT-SIP

destination-pattern .T

progress_ind progress enable 8

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.205.20.50:5060

session transport udp

voice-class codec 1

dtmf-relay rtp-nte digit-drop

no vad

!

dial-peer voice 30 voip

incoming called-number .

description To CallManager - SBWPMPUB

destination-pattern [23]...$

session target ipv4:192.168.12.190

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

dial-peer voice 31 voip

description to Callmanager-subscriber

incoming called-number .

preference 1

session target ipv4:192.168.12.189

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

HTH

Anas

please rate if it is helpful