04-01-2013 06:50 AM - edited 03-16-2019 04:32 PM
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
04-01-2013 07:03 AM
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
04-01-2013 07:09 AM
thankyou Mr.Chris,
Tried the commands but still the call ringing at ext 2300 not going to 3080
04-01-2013 07:12 AM
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
04-01-2013 07:23 AM
04-01-2013 02:48 PM
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
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