05-28-2019 08:53 AM
Hello,
From our SIP trunk we are stripping down to just the last 5 digits of each incoming call. If the last 5 digits of the DID being called doesn't need to translate to a different extension we don't have any translation configured. One situation we have run into, and due to my lack of experience with CUBE routers, is we have a DID for example 123-456-7890 that needs to translate to 67990, not 67890. If I have an already configured DN of 67890 then the call goes to that extension and doesn't seem to match on my CUCM translation and send the call to configured DN 67990. My understanding is that what I need to do is translate the call from the CUBE router. From my research I found that the configuration on the CUBE should look something like the following. However, the call doesn't translate from the CUBE. I seem to have done something incorrectly.
voice translation-rule 100
rule 1 /1234567890/ /67990/
rule 2 /1234567892/ /67992/
voice translation-profile Inbound_Xlate
translate calling 100
dial-peer voice 79001 voip
description cube-to-ucm-std
translation-profile incoming Inbound_Xlate
session protocol sipv2
session server-group 10
destination e164-pattern-map 7901
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
codec g711ulaw
Thanks in advance for your help, and time!
Solved! Go to Solution.
05-28-2019 02:13 PM
I'd strongly recommend you read this:
You need to understand the concept of inbound and outbound dial peers, and then read:
05-28-2019 09:28 AM
You're translating the calling number, not the called number.
Based on the signaling type of the incoming call, the calling number is equivalent to Automatic Number Identifier (ANI) or the calling line id. The redirect-called number is equivalent to redirect Dialed Number Identification Service (DNIS) or the original called number.
05-28-2019 10:19 AM
Hi Jaime,
Thanks for the quick response. So all I will need to do is make the below changes to my config and it should work?
voice translation-profile Inbound_Xlate
translate called 100
Thanks
05-28-2019 01:08 PM
java,
I changed my config to the below configuration but the call still does not translate to the desired DN.
voice translation-profile Inbound_Xlate
translate called 100
Thanks again for the help!
05-28-2019 01:19 PM
Then debug your CUBE to make sure you're actually hitting the dial peers you want to.
And to confirm you're actually getting the digits you're using for the translation from the provider.
05-28-2019 01:48 PM
So I ran a debug voice ccapi inout and I got the following information. This tells me that it did hit my 79001 dial-peer. So what am I missing? *I changed information in the debug for privacy reasons*
ccCallSetupRequest:
cisco-username=1234561234
----- ccCallInfo IE subfields -----
cisco-ani=1234561234
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=1234567890
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
2272125: *May 28 17:36:27.056 EDT: //1231811/7B830D7C9DCF/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x267F6A64, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=1234561234,(Calling Name=XXXXXXXX)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=1234567890(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=79001, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Dial-peer config:
dial-peer voice 79001 voip
description cube-to-ucm-std
translation-profile incoming Inbound_Xlate
session protocol sipv2
session server-group 10
destination e164-pattern-map 7901
voice-class sip bind control source-interface Loopback1
voice-class sip bind media source-interface Loopback1
dtmf-relay rtp-nte
codec g711ulaw
05-28-2019 02:13 PM
I'd strongly recommend you read this:
You need to understand the concept of inbound and outbound dial peers, and then read:
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