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Robert Craig
Participant

Weird caller id

I am messing around in a lab with two differen CUCM's. Site A and Site B both have different area codes to lab two different states essentially. Calling between the two works great over a sip trunk. I setup a simple 10 digit route pattern between the two (wild card) and they can call each other. The DN's are 7 digits, but I have their external phone number mask set as 520XXXXXXX or 760XXXXXXX. I have a translation pattern on each side that strips the area code off of the incoming calls (760.XXXXXXX and 520.XXXXXXX). The problem is, when either site calls each other, the displayed caller id is missing the area code. Example:

Site A --->>> Site B

While the phone is ringing on B, the Called Number on A only shows 7 digits instead of 10. Site B phone (still ringing) shows an incoming call from A with the full 10 digits. I can repeat this process in the opposite direction with the same results. What I can't get my head wrapped around is why the Calling phone won't display the full 10 digits that I just dialed. Anyone have any suggestions? 

16 REPLIES 16
Roger Kallberg
VIP Advisor

Let me see if I understood this correct. Your phones use 7 digit DNs and you have set the EPNM to show the full 10 digit "DID DN". Then you have a RP that points the 10 digit of the remote site over the trunk.

On that RP have you set use EPNM for the calling number transformation? If not it will only show the actual DN that the caller have, aka the 7 digit that you have set on the line.

EPNM will not be used if you don't select it on a RP or TP. The only use of the mask is then to affect the displayed number on the top right of the phone display. EPNM have other use also, like AAR, but that's outside the scope of your question.

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Selected it, no effect.

Site A >>>>> Site B

Site A Phone >>> [2-9]XXXXXXXXX (use EPNM checked] >>>>> SIP to Site B >>>>> (Site B) TP 760.583XXXX (predot) >>> Site B Phone

Site A displays calling 5831000 (actual digits dialed 7605831000) ---- Site B displays incoming call from 5205331000

Ok, now the penny dropped. It's the displayed called number on the caller phone that shows the 7 digits. I think thats because of remote CUCM signaling back to the originating CUCM that the called number has been modified. Not 100% sure about SIP, but this is the case with H.323 anyway.

You could try to modify the called number on the originating CUCM before it's sent to the SIP trunk. To do that without affecting the display you need to do the DDI on the route list/route group details. If you do it on the RP itself it will affect what is shown on the display of the caller.

There is also a service parameter that can be set to always show originally called number, but that will affect all calls within the cluster, so it might break something else for you.

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OK, I'm having a hard time following you. I created a RG and RL and put that one trunk in it, but where would I configure anything involving digits?

On the RL/RG details page.

After you have selected the RG as a "member" of the RL you will have it at the bottom of the page. Click that link and you'll get to the details page were you can do digit manipulation on called and calling number.

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Yup, got it. Forgot about that screen. I check EPNM to ON, but no result. Should I append the area code, or do a digit strip on that page instead of a translation pattern?

Do the digit strip on that page instead of using a TP on the far end. Also insure that the incoming CSS on the SIP trunk can see the PT that holds the 7 digit DN.

The use EPNM only affect what is sent as the calling number to the receiving end, not what is shown as called number on the caller phone.

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I've never used this page before. Unlike TP, I can't put in 760.583XXXX. It is rejecting the DOT, even though I can select NANP:PreDOT. I am configuring this on Site A. I deleted the TP on Site B, which was stripping 760. Now calls going from A to B are failing because 10 digits are going to B, but nothing is stripping it.

Roger Kallberg
VIP Advisor

You need to change you RP so that it has the . (dot) in the correct place in the match statement. Then you do a DDI=predot on the RL/RG details page.

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OK, I see what you are saying. However, this is between two clusters. What I wanted to simulate was "long distance" calling from one cluster to another. Yes, I know there is no ITSP in between, but shouldn't this apply? If the RP went to an ITSP, the ITSP would only receive 7 digits, right?

OK, so I changed [2-9]XXXXXXXXX to [2-9]XX.XXXXXXX on both sides, as well as under the RG details, under Called Number, selected NANP:PreDOT. Calls go through, but still back to 7 digits displayed on Calling phone.

Roger Kallberg
VIP Advisor

Hmm tricky problem, could you try to do this. On the RP do a DDI=predot and prefix the area code. That should affect what is seen in the display of the caller.. So basically you remove the area code with predot, then add it back with the prefix. Test this first with the DDI you have on the RL/RG so that it sends 7 digits.

If that works and if you then want to send the full 10 digit to the remote site and use the TPs that you had from the start do no DDI on the RL/RG and add the TPs back on each end in the call path.

That might of course take you right back to were you started, but it's worth a try.

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Yeah, still same problem. No matter how I did it with appending, stripping, at whatever level, it still only displayed calling seven digits. I ended up just using 10 digit DN's. I just can't believe it's this hard. Do most public environments use 10 digit DN's or do what I was attempting to do in the beginning with usign seven and appending the area code when egressing to the ITSP?

I have not used ITSP connections that much, but I would guess that your setup is not fully comparable with a real world scenario. Most ITSP connections would go through a CUBE, and there you can turn off the update of called number.

On a H.323 to SIP CUBE you would use this command.

voice service voip

no sup serv h225 upd

And on a pure SIP CUBE you would use this.

voice service voip

sip

no display-caller update

Not 100% sure about the last command, but it's something similar.

You don't happen to have a GW that you can use as a CUBE, about any router would do, so that you can route the calls via that?

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