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989
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Voice translations help

I am trying to setup a simple voice translation to dial 9 out....

We dial our numbers as: 9+1+xxx+xxx+xxxx

We need 9 stripped so that way1XXXXXXXXXX is pushed to our SIP trunk.

I tried a few ways and it goes busy.

Also what is the destination pattern supposed to be in conjunction with this?

7 Replies 7

Dennis Mink
Advisor
Advisor

I dont think you need a translation pattern,  If you want to route 91XXXXXXXXX   just add a route pattern on CUCM 9.1XXXXXXXXX  and discard digits predot and point that to your TRunk as you would with any Route pattern.

(assuming you are doing this on a CUCM)

Please remember to rate useful posts, by clicking on the stars below.

Anas Abueideh
Engager
Engager

Hi,

if you want just to remove the 9 you can apply the following config

voice translation-rule 1

  rule 1 /^9\(.*\)/ /\1/

voice translation-profile discard9

  translate called 1

at the outbound dial-peer you need to assign the translation profile

dial-peer voice 100 voip

destination-pattern 91.........

translation-profile outgoing discard9

HTH

Anas

please don't forget to rate the helpful posts

So can you explain why its called not calling?

I have attached my config, for better way to associate any changes...

What we would like to do is setup:

Dial 9 to get out and the remaining digits for local

Dial 9 to get out, with 1 for national calls

911 calling

411 calling

Could you assist in giving us a config for us to use?

Hi Frederick,

we the translation profile for the called because we need to manipulate the called number and remove 9.

for 911 calls you don't need to apply the translation profile.

you need to create dial-peers for the different route patterns.

for calling number you need to create translation-profile to change the calling number to your company DID

HTH

Anas

please don't forget to rate the helpful posts

your config should be like the following

voice translation-rule 1

rule 1 /7347947685/ /1000/

!

voice translation-rule 2

  rule 1 /^9\(.*\)/ /1/

voice translation-rule 3

   rule 1 // /7347947685/

!

!

voice translation-profile IN_TO_1000

translate called 1

!

voice translation-profile SIP_OUTBOUND

translate called 2

translate calling 3

dial-peer voice 2000 voip

description **VoIP MS OUTBOUND**

translation-profile outgoing SIP_OUTBOUND

destination-pattern 900T

session protocol sipv2

session target sip-server

session transport udp

voice-class sip asserted-id ppi

no voice-class sip block 180

no voice-class sip block 183

no voice-class sip block 181

voice-class sip pass-thru headers unsupp

voice-class sip pass-thru content unsupp

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec g711ulaw

no vad

dial-peer voice 2001 voip

description **VoIP MS OUTBOUND**

destination-pattern 911

session protocol sipv2

session target sip-server

session transport udp

voice-class sip asserted-id ppi

no voice-class sip block 180

no voice-class sip block 183

no voice-class sip block 181

voice-class sip pass-thru headers unsupp

voice-class sip pass-thru content unsupp

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec g711ulaw

no vad

dial-peer voice 2002 voip

description **VoIP MS OUTBOUND**

destination-pattern 411

session protocol sipv2

session target sip-server

session transport udp

voice-class sip asserted-id ppi

no voice-class sip block 180

no voice-class sip block 183

no voice-class sip block 181

voice-class sip pass-thru headers unsupp

voice-class sip pass-thru content unsupp

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec g711ulaw

no vad

dial-peer voice 2004 voip

description **VoIP MS OUTBOUND**

translation-profile outgoing SIP_OUTBOUND

destination-pattern 91.......$

session protocol sipv2

session target sip-server

session transport udp

voice-class sip asserted-id ppi

no voice-class sip block 180

no voice-class sip block 183

no voice-class sip block 181

voice-class sip pass-thru headers unsupp

voice-class sip pass-thru content unsupp

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec g711ulaw

no vad

dial-peer voice 2004 voip

description **VoIP MS OUTBOUND**

translation-profile outgoing SIP_OUTBOUND

destination-pattern 9.......$

session protocol sipv2

session target sip-server

session transport udp

voice-class sip asserted-id ppi

no voice-class sip block 180

no voice-class sip block 183

no voice-class sip block 181

voice-class sip pass-thru headers unsupp

voice-class sip pass-thru content unsupp

voice-class sip pass-thru content sdp

dtmf-relay rtp-nte

codec g711ulaw

no vad

HTH

Anas

please don't forget to rate the helpful posts

Got it working thanks!   Now if I can only get MWI working with exchange, any ideas?

Hi,

what is your MWI issue with exchange ?

Anas

please don't forget to rate the helpful posts

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