Hello,
Calling Number is 1209
Called Number is 9XXXXXXXXXX
I want to translate Calling Number to 0312XXXXXXX and Called Number to 05XXXXXXXXX but translation-rules on the translation profile doesn't work with below configuration. rule 9 should be used for the called number and delete prefix 9 and rule 12 should be used for calling number and the calling number should be translated to the our customer's number that's registered to ISP but it is not working.
Can you help me to find my mistake on this configuration please
Also the call scenario is below.
Webex Calling --> SIP --> CUBE -->FXO --> ISP
The CUBE was configured for Webex Calling with on Premises PSTN.
voice service voip
ip address trusted list
ipv4 23.89.0.0 255.255.0.0
ipv4 85.119.56.0 255.255.254.0
ipv4 85.119.57.128 255.255.255.192
ipv4 128.177.14.0 255.255.255.0
ipv4 128.177.36.0 255.255.255.0
ipv4 135.84.168.0 255.255.248.0
ipv4 135.84.169.0 255.255.255.128
ipv4 135.84.170.0 255.255.255.128
ipv4 135.84.171.0 255.255.255.128
ipv4 135.84.172.0 255.255.255.192
ipv4 135.84.173.0 255.255.255.128
ipv4 135.84.174.0 255.255.255.128
ipv4 139.177.64.0 255.255.248.0
ipv4 139.177.64.0 255.255.255.0
ipv4 139.177.65.0 255.255.255.0
ipv4 139.177.66.0 255.255.255.0
ipv4 139.177.67.0 255.255.255.0
ipv4 139.177.68.0 255.255.255.0
ipv4 139.177.69.0 255.255.255.0
ipv4 139.177.70.0 255.255.255.0
ipv4 139.177.71.0 255.255.255.0
ipv4 139.177.72.0 255.255.255.0
ipv4 139.177.73.0 255.255.255.0
ipv4 144.196.0.0 255.255.0.0
ipv4 150.253.128.0 255.255.128.0
ipv4 163.129.0.0 255.255.128.0
ipv4 170.72.0.0 255.255.0.0
ipv4 170.133.128.0 255.255.192.0
ipv4 185.115.196.0 255.255.252.0
ipv4 185.115.197.0 255.255.255.128
ipv4 199.19.196.0 255.255.254.0
ipv4 199.19.197.0 255.255.255.0
ipv4 199.19.199.0 255.255.255.0
ipv4 199.59.64.0 255.255.248.0
ipv4 199.59.65.0 255.255.255.128
ipv4 199.59.66.0 255.255.255.128
ipv4 199.59.67.0 255.255.255.128
ipv4 199.59.70.0 255.255.255.128
ipv4 199.59.71.0 255.255.255.128
ipv4 62.109.251.41
media statistics
media bulk-stats
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
trace
stun
stun flowdata agent-id 1 boot-count 5
sip
early-offer forced
g729 annexb-all
audio forced
!
!
voice class uri 200 sip
pattern dtg=XXXXXXXXXXXX
voice class codec 99
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
codec preference 4 g729r8
codec preference 5 opus
codec preference 6 g722-64
!
voice class codec 1
codec preference 1 g729br8 bytes 30
codec preference 2 g729r8
codec preference 3 g728
codec preference 4 g711alaw
codec preference 5 g711ulaw
!
voice class stun-usage 200
stun usage firewall-traversal flowdata
!
!
voice class sip-profiles 200
rule 9 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 10 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 11 request ANY sip-header From modify "<sips:" "<sip:\1"
rule 12 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 13 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 14 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 15 response ANY sip-header Contact modify "<sips:(.*)" "<sip:\1"
rule 20 request ANY sip-header From modify ">" ";otg=XXXXXXXXXXXX>"
rule 30 request ANY sip-header P-Asserted-Identity modify "sips:(.*)" "sip:\1"
!
!
!
voice class custom-cptone Hat_Kapama
dualtone disconnect
frequency 450
cadence 200 200 200 200 200 200 600 200
!
voice class tenant 200
registrar dns:XXXX scheme sips expires 240 refresh-ratio 50 tcp tls
credentials number XXXXX username XXXXXXXXX password 6 XXXXXXXXXXXX realm BroadWorks
authentication username XXXXXXXXX password 6 XXXXXXXXXXXX realm BroadWorks
authentication username XXXXXXXXX password 6 XXXXXXXXXXXX realm xxxxxxxx
no remote-party-id
sip-server dns:xxxxx
connection-reuse
srtp-crypto 200
session transport tcp tls
url sips
error-passthru
asserted-id pai
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
no pass-thru content custom-sdp
sip-profiles 200
outbound-proxy dns:xxxxxxxxxxxxxx
privacy-policy passthru
!
voice class tenant 100
session transport udp
url sip
error-passthru
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
no pass-thru content custom-sdp
!
voice class tenant 300
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
no pass-thru content custom-sdp
!
voice class srtp-crypto 200
crypto 1 AES_CM_128_HMAC_SHA1_80
!
!
!
voice register global
default mode
no allow-hash-in-dn
system message Yedek Sistem Devrede
max-dn 100
max-pool 100
!
voice register pool 1
id network xxx.xxx.xxx.0 mask 255.255.255.0
number 1 69.. preference 10
preference 10
dtmf-relay rtp-nte sip-notify
!
!
voice translation-rule 9
rule 1 /^9/ //
!
voice translation-rule 12
rule 1 /12../ /XXXXXXXXXXX/
!
!
voice translation-profile DISARAMA
translate calling 12
translate called 9
!
!
interface GigabitEthernet0/0/0
ip address XXXXXXXXXXXXXXXXXXXXXX
negotiation auto
!
interface GigabitEthernet0/0/1
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/3
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/4
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/5
no ip address
shutdown
negotiation auto
!
interface Service-Engine0/1/0
!
interface Service-Engine0/2/0
!
ip http server
ip http authentication local
ip http secure-server
ip http client source-interface GigabitEthernet0/0/0
ip forward-protocol nd
!
!
!
!
!
!
!
control-plane
!
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone Hat_Kapama
no vad
cptone TR
timeouts interdigit 5
timeouts call-disconnect 5
timeouts wait-release 5
timing hookflash-out 50
timing guard-out 1000
connection plar 1200
caller-id enable
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 10 pots
description *** GSM Calls ***
preference 1
destination-pattern 05.........
port 0/1/0
forward-digits 11
!
dial-peer voice 11 pots
preference 1
destination-pattern 1..
port 0/1/0
forward-digits 3
!
dial-peer voice 12 pots
description *** International Calls ***
preference 1
destination-pattern 00T
port 0/1/0
forward-digits all
!
dial-peer voice 13 pots
description *** Local Calls ***
preference 1
destination-pattern [2-8]......
port 0/1/0
forward-digits 7
!
dial-peer voice 14 pots
description *** Long Calls ***
preference 1
destination-pattern 0[2-4].........
port 0/1/0
forward-digits 11
!
dial-peer voice 15 pots
description *** 08XX Calls ***
preference 1
destination-pattern 08.........
port 0/1/0
forward-digits 11
!
dial-peer voice 100 voip
description *** WEBEX ***
translation-profile incoming Outbound
translation-profile outgoing test
destination-pattern 12..
session protocol sipv2
session target sip-server
voice-class codec 99
voice-class stun-usage 200
no voice-class sip localhost
voice-class sip tenant 200
dtmf-relay rtp-nte
srtp
no vad
!
dial-peer voice 101 voip
description ***From WEBEX ***
translation-profile incoming DISARAMA
max-conn 250
session protocol sipv2
incoming called-number .
voice-class codec 99
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte sip-kpml sip-notify
dtmf-interworking standard
srtp
no vad
!
