connecting elastix (freebpx based, witch is based on asterisk) to omnipcx enterprise

[am1]

Here it is:

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x x
x Node Number (reserved) : 1 x
x Instance (reserved) : 1 x
x Instance : 0 x
x x
x Gateway Name : Elastix x
x Remote domain : 10.10.1.24 x
x PCS IP address : ----------------------------------------------- x
x Port number : 5060 x
x Transport type + UDP x
x Belonging domain : -------------------------------------------------- x
x Registration Id : -------------------------------------------------- x
x Registration ID in P_Asserted + False x
x Registration timer : 0 x
x Outbound Proxy : -------------------------------------------------- x
x Supervision timer : 0 x
x Trunk group number : 11 x
x Pool Number : -1 x
x Outgoing realm : -------------------------------------------------- x
x Outgoing username : -------------------------------------------------- x
x x
x Outgoing Password : ---------- x
x Confirm : ---------- x
x x
x Incoming username : -------------------------------------------------- x
x x
x Incoming Password : ---------- x
x Confirm : ---------- x
x x
x RFC 3325 supported by the distant + True x
x DNS type + DNS A x
x First DNS IP Address : ----------------------------------------------- x
x Second DNS IP Address : ----------------------------------------------- x
x SDP in 18x + True x
x Minimal authentication method + None x
x INFO method for remote extension + False x
x Send only trunk group algo + False x
x To EMS + False x
x Routing Application + False x
x Ignore inactive/black hole + False x
x Contact with IP address + False x
x Dynamic Payload type for dtmf : 97 x
x 100 REL for Outbound Calls + Not Supported x
x 100 REL for Incoming Calls + Not Requested x
x x
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Please forgive the noob question but as I said I have little experience with this system. How do I exactly go on collecting call traces for a certian number on the oxe side?

[vad]

I cannot select and change the Q931 signal variant for some reason...
It goes from Private Trunk Group to SS7 signal variant skipping the line in the middle.
Am I doing something wrong?

Edit: Even if I change the DID when I outbound dial to 110 and the call from OXE reaches the other side of the trunk, I cannot see any "Destination Number" DID in the sip header on the other side.
It just says To: <sip:110@10.10.1.24;user=phone>
How can I fix this?

We have OUTGOING call from OXE?
Destination - we talk about calling or called number?
I think (before) that 1xx - users in OXE (and we talk about DID translator and CALLING number).
But as I understand now - 1xx - users in Elastix. You dial 110 (network number - user on another site, not OXE) and OXE calls 110 via SIP gateway 0.
In this case - not interesting DID rules (DID rules used during INCOMING call to OXE - called number analyzing, and during OUTGOING call - for CALLING number building).
Explain:
1xx - user in OXE or Asterisk site
5xx - user in OXE or Asterisk site
what need to dial OXE user
what need to send OXE as called number (called destination)

[am1]

Yes, 110 is in Elastix.
It works like this:
Dial 110 in OXE -> OXE Trunk 11 -> Elastix
I see the call incoming on Elastix so the trunk is working.
"From" calling number is correctly recognized.
The only problem is that in the SIP Header received by Elastix the Called number DID is missing in the To (Called number part).

Here is what OXE currently sends:
To: <sip:110@10.10.1.24;user=phone>

It's missing an identifier for the other end of the trunk.
This means that I cannot route the call on the other side.

I need OXE to send DID for number 110 in the SIP Header or maybe a "display name".
For example what I need is:
To: "155154646441" <sip:110@10.10.1.24;user=phone>

Any way to do that?