calls :404 not found from CUCM

[akramroot]

Hello all,

I have this scenario :

Server (PBX) connected  on cucm via sip trunk , the cucm connected via PSTN sip trunk for external calls.

PBX ---- CUCM ----PSTN 

-When a user from PBX call a normal PSTN numbers  the call is passed via cucm after it arrive at PSTN  and function correctly .

- When the same user call an PSTN anonymous nmbers  the call is rejected by CUCM with termination cause code (1) Unallocated (unassigned) number  

When i checked the normalized message i found SIP /2.0 404 no found , and on invite message:
From: "PBX_GW" <sip:PBX_GW@adress ip of PBX>

The cucm send the 404 not found to PBX.

Do you have any idea of this problem ? does the cucm not know the format sip:PBX_GW@adress ip of PBX ?

Could you help me please 

Best regards

[Maren Mahoney]

What do you mean when you say "When the same user call an PSTN anonymous numbers" (What do you mean by "anonymous numbers"?)

When the PBX sends the call to the CUCM, the Inbound Calling Search Space on the SIP Trunk must contain some kind of pattern to capture and process the "To" number in the SIP Invite. When these '"anonymous numbers' are dialed, does the trunk in CUCM have a route pattern or translation pattern that is able to match the number?

If not, there is your issue. CUCM would issue a 404 back to the PBX because it is unable to 'find' a matching pattern.

If CUCM does have a match, but is unable to route the call to the PSTN successfully it is possible that it is a downstream system that is unable to process the call - and that system sending its error back to CUCM causing CUCM to generate a 404 back to the PBX. I would expect CUCM to generate a 503 in this scenario (which is "I recognize the number dialed but am unable to process the call'), but I could see a scenario where it generates a 404 instead.

If you can't logically determine if CUCM is able (or not) to process an 'anonymous' call sent by the PBX, you could use the Dialed Number analyzer (choose "Trunk" from the menu) to see what CUCM is thinking when it receives the inbound SIP Invite information.

Maren 

[akramroot]

Hi  @Maren Mahoney , many thanks for your time and your help .

When the same user call an PSTN anonymous numbers" (What do you mean by "anonymous numbers"?) i mean  the user made a hidden call .(private call)

How can i do the test by DNA : 

I have analyzer input (dircetory URI , or calling party ) and dialed digit settings (directory URI , or dialed digits)

Thank you in advance 

Best regrads 

[Roger Kallberg]

To add to what @Maren Mahoney wrote. CM doesn’t route on information in the From header normally. It uses the information in the received invite URI field to route calls.

[akramroot]

Hi @ Roger Kallberg , thank you for your helep 

This is the call flow that i have :

The detailled sip message on 404 not found  i have :

from :"PBX_GW" <sip:PBX_GW@adress_ip_of_PBX>

to :<sip:+33654338940@adress_ip_of_cucm>

Reason: Q.850;cause=1

Best regards 

[Roger Kallberg]

Exactly what are you expecting anyone to do with this limited information? If you want help from the community you’ll need to provide actual data that contains the call setup SIP dialogue. Without that we’re just guessing. You’ve been around long enough in the community to know that as I and a number of peers have answered multiple questions from you in the past.

[Roger Kallberg]

Cause code 1 means this.

Cause 1 Unallocated (unassigned) number - This cause indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned).

[Maren Mahoney]

So first, with the DNA, as I wrote earlier you will want to select "Trunk" from the DNA menu. That allows you to choose the trunk on which the call is incoming and enter the information from the INVITE. My guess is, based on the 404 error, that the DNA will not find a path for the call in CUCM.

So, look again in CUCM at the Inbound Calling Search Space configured on the trunk to the PBX. Does the Inbound Calling Search Space have a partition that has a pattern that will match the +33654338940 number dialed? The 404 error would indicate that there is no pattern available to the trunk that matches those dialed numbers. You can test this by creating a route pattern with the above number exactly, and putting it in a partition available to the trunk and trying the call again.

Maren

[akramroot]

Thank you @Maren Mahoney  for your help .

When the user call from PBX to PSTN > the call passed without any problem 

In the invite : from : <sip:0655893340@adress ip of pbx>

to : <sip:0654338940@ip of cucm>

all is good .

but  when he calls with a hidden number the call not passed 

in the invite :

from :"PBX_GW" <sip:PBX_GW@adress_ip_of_PBX>

to :<sip:0654338940@adress_ip_of_cucm>

I would like to know whey  "PBX_GW"  showing instead of number "0655893340"

is this normal behavior ?

Best regards

[Roger Kallberg]

Why it shows that in the From field is a question that you should address to those that administer the PBX. 

[Roger Kallberg]

By any chance have you looked at if you have this setting set to something else than Off?

 

[akramroot]

Thank you @Roger Kallberg  for your return and your help .

I checked that : anonymous call block  is disabled on on SIP Profile .

Best regards 

[Maren Mahoney]

What sort of PSTN connection do you have? Is it also SIP or is it something else? If it is a PRI, is it possible that CUCM is signaling to the PRI and the call is rejected there?

Maren

[akramroot]

Thank you @Maren Mahoney  for your return,

The PSTN connecivity with a sip trunk SBC.

Best regards

[Roger Kallberg]

Have you done the DNA for the SIP trunk as @Maren Mahoney asked you to do? It would be interesting to see what it matches for the two call cases.