Unassigned DID numbers going to one extension

[robandover385]

We are having an issue where we have unassigned DID phone numbers that all end up at one directory number, but then if we setup a translation patterns for that DID, then it routes to the directory we just assigned it to. We are trying to see if there is a way to have all unassigned DID numbers go to a not in service voice recording. We are running CUCM 12.5 and our voicegates are Cisco 4351 using SIP trunks. I know we have voice class pattern-maps configured on the voice gateways, as e164 ^3162184[2-8]..$ but don't know where those are getting pointed at in CUCM. I have searched for these numbers in the route plan report but they don't show up. 

[Roger Kallberg]

The SDL traces are also known by Call Manager traces. It includes details on the call processing. On the debugs it depends on what type of connection you have towards PSTN. As a start you can enable debug ccsip message and debug voip ccapi inout as you have a SIP trunk towards CM.

[robandover385]

Thanks, I'll start with thoes. How would I be able to find out the type of connecttion we have towards our PSTN? I know all of phones are VOIP but then looking at the gateway our interfaces have media-type rj45 on gigabit ports.

[Roger Kallberg]

You will see that in the configuration in the router. Please share the running configuration if you don’t know how to.

[Roger Kallberg]

You have a SIP trunk to PSTN. Not related to your question directly, but I would recommend that you do these changes to your configuration.

!**Reference info**
!voice class server-group 1
! ipv4 172.17.XXX.XX preference 2
! ipv4 172.17.XXX.XX preference 1
! description *** CUCM Servers ***
!!
!voice class server-group 2
! ipv4 198.241.XX.XXX preference 2
! ipv4 198.241.XX.XXX preference 3
! description *** IdeaTek SIP Servers ***
!**Reference info**


!Take all the IPs from the voice class server-group 1 and put into this, if you have more CPE CM nodes add those as well
voice class uri CUCM sip
 host ipv4:172.17.XXX.XX
 host ipv4:172.17.XXX.XX

!Take all the IPs from the voice class server-group 2 and put into this
voice class uri PSTN sip
 host ipv4:198.241.XX.XXX
 host ipv4:198.241.XX.XXX

!Use information in the VIA header to match the inbound direction
dial-peer voice 100 voip
 no incoming called e164-pattern-map 1
 incoming uri via PSTN
!
dial-peer voice 200 voip
 dtmf-relay rtp-nte sip-kpml
 no incoming called e164-pattern-map 2
 incoming uri via CUCM

And as well remove any MGCP configuration that you have as you don't seem to be using that at all based on the shared configuration.

[robandover385]

Ok, I'll look into those changes but I will admit that I am a networking guy and not a phone system expert. Do you see anything on the GW that could be doing the routing. I also ran the RTMT and looked up the number this was happpening to and seems like it got an invite, 100 trying, then a 404 not found, and final an ACK. I can get the full message details if needed.

[Roger Kallberg]

There is from what I could tell nothing in the GW that would explain how the call lands on one specific directory number. I think you'll need to take the bull by the horns and look at the Call Manager logs (SDL traces). Being a network guy you're in for a steep learning curve. Those files are not directly easy to read when you look at them for the first time.

[robandover385]

If I send over some the messages from the anaylze call diagram, are you able to give me a little insight?

[Roger Kallberg]

Better if you where to download the SDL files and share them. You do that from RTMT. The call diagram will not show you much more than you already know.

[robandover385]

What is SDL short for? Can you share with how I can download the SDL files from RTMT so I can share them? Sorry, I know I am asking a lot, just trying to get my head wrapped around everything and learn in the process.

[Roger Kallberg]

You download the files from Collect Files in RTMT. If you search for download trace/log files from CUCM you’re going to find plenty of information about this.

SDL stands for Signal Distribution Layer.

[robandover385]

So I was able to download the files and attached is the zipped folder.

[Roger Kallberg]

Can you please share these details so that we can find the call in question.

• called number, ie the unassigned number that was called 
• calling number 
• time of the call 

[robandover385]

called number- 316-218-4591

calling number- 316-494-2895

time of call- 2:54 pm

[Roger Kallberg]

None of the files you sent include any call that matches that.

The first entry in the files are from 13:46:33.381 and the last is from 13:59:31.286, so it doesn't seem to include the time of your call and in fact the files does not contain any calls as seen below.

One advice, check the trace settings for the call manager service, it should look something similar to this.

Redo the call, note down the time of when you did the call. Download new trace files and also get a tool called TranslatorX, you can get it from here https://www.translatorx.org/  and open the downloaded file, once extracted if you select to have the compress file option set in the Collect Files in RTMT, which I would recommend you to not have, you can just drag and drop the entire folder to the UI of TranslatorX.