SIP CUBE Header Manipulation to read DNIS

[Hieronymous Merkin]

I have a CUBE setup for SIP trunk to ITSP. It works and all is good. The issue is that we forward DNIS numbers at a remote location to an 800 number that terminates on this QUBE. We need to see the DNIS or CallED number dialed by the caller, that was forwarded to this toll free number. Not the CallerID, but the number the caller Dialed. The calls are completing but follow the dial-peers for the 800 number. How do we see the DNIS so we can create routing by the number dialed?  Not sure what this would be referred to as.

All comments, recommendations and advice appreciated!  Please point me to configuration examples. Thanks.

[Evgeny Izetov]

Is this CUCM or CME? Do you have Unity Connection? How many DID numbers from the remote site are being forwarded?

I'd first check that the ITSP actually delivers the redirecting number. Run "debug ccsip messages" and look for Diversion Header in the INVITE.

I can think of one way of accomplishing what you want with Unity Connection but it's not scaleable.

[Hieronymous Merkin]

This is about ITSP connection to a CUBE router.  This really will not impact Unity, it will impact DNIS in the UCCX however.   We need to see the number the caller dialed, that was forwarded to the inbound toll free number.   We see the toll free number, but not the number that was originally called. 

[Evgeny Izetov]

Check that the PSTN provider sends the Diversion Header first. Then see that on the SIP trunk to CUBE in CUCM you have checked "Redirecting Diversion Header Delivery - Inbound". Then in UCCX you may be able to pull that number in Get Call Contact Info step as Original Called Number or Last Redirected Number.

[DrVoIP · Accepted solution]

Configuration to match incoming dial-peer using TO header username:-

Conf t

voice translation-profile TEST

translate called 101

voice translation-rule 101

rule 1 /^\(8442158437\)/ /ABCD/

voice class uri 101 sip

user-id 9313645321

incoming DP:-

dial-peer voice 101 voip

translation-profile incoming TEST

session protocol sipv2

incoming uri TO 101

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

Configure outbound dial-peer accordingly:-

dial-peer voice 1001 voip

description outbound to CUCM

preference 1

destination-pattern ABCD$

session protocol sipv2

session target ipv4:10.10.12.10

voice-class codec 1 

 voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

end