CUCM connected to the PSTN via SIP. On the trunk in the call manager we strip all incoming down to 5 digits (significant digits = 5) as we use a 5 digit dial plan for all phones. We are now porting 3 DID ranges that will be routed off to a sip trunk for our faxmaker server. These 3 ranges overlap current numbers. What I would like to do is to allow these 3 ranges to come in with their full 10 digits so that I can route these ranges independently.
We currently have a plan to remove the 5 digit strip in the call manager and strip using dial peers in the cube, and then add dial peers for each of the DID ranges to not strip and make them a higher preference.
Is this the correct path or is there a better way to accomplish this?
So what is the DNIS on the incoming from the PSTN via SIP (that you are currently stripping to 5)? I assume 10 digits?
I would take the strip off the trunk and let 10 digits ingress for everything, then use translations/transformations/css/partitions to do your striping. Anything matching the 3 DID ranges goes to the fax server trunk, anything else gets stripped to 5 digits.
Just sounds like a lot of work messin' with dial peers .... etc
I am assuming that the strip happening on the trunk is before the translation pattern match?
Trying to wrap my head around what a translation pattern looks like to strip all incoming to 5 digits.
Sounds like I need to remove the trunk strip, create translation pattern to strip all incoming to 5 digits and route normally, and then create translation patterns for each of the overlapping DID ranges to route these to fax sip trunk.(I assume I do this by creating a called party transform mask? Does the original ANI continue? I would need this as it is a fax call.)
Assuming all DNIS is 10 digits, I would let all 10 digits ingress.
Create a partition called Ingress_PT and a CSS called Ingress_CSS (the Ingress_PT is the only partition in Ingress_CSS).
Create a translation pattern in Ingress_PT with the pattern ! using a CSS that only talks to the partition all the DNs/endpoints are in. Use the Called Party Transformation mask to strip the 10 digit number down to 5 digits (XXXXX would strip to the last 5 digits).
Assign Ingress_CSS as the CSS for your ingress SIP trunk from the ITSP.
This should do essentially what you are doing now; stipping all ingress called party numbers down to the last 5 digits.
Create the route patterns/lists/groups to your SIP trunk for the fax server and place them in Ingress_PT. Your route patterns will need to match your 3 DID ranges as specifically as possible. For example, 1419503XXXX
The reason this works is due to how call manager processses route matches; with call manager the most specific match always wins the routing decision. So, 1419503XXXX (for example) is always more specific than !. Anything matching 1419503XXXX will follow the rules (and routing) of that pattern, everything else would match ! and get stripped to 5 digits and ingress to the partition where all the endpoints are.
All ANI and header info from the carrier should remain intact. Depending on your system defaults though, you may need to allow the Calling Line ID and Name Presentation.
As always, I would LAB this up before pushing to production.