Call failing on the SIP trunk disconect cause 404

[yamikani2g2]

Good day Experts 

I have this issue, I have 100 lines for my office coming in an E-1 line, in this i have some number that are not in a sequence that have to be recorded. I got a 3rd party PBX. The experts in here advised that i have a SIP trunk to my CUCM and SIP trunk to MyPBX.

The team here has been helpful so far so good the SIP trunk between the Cisco-Gateway and MyPBX works like a Charm. I now have an issue with the second SIP trunk to CUCM. I don't seem to be getting the second SIP trunk working.

Below are my configuration and attached are my debug Outputs.

voice translation-rule 8248

 rule 1 /^367389$/ /8248200/

!

!

voice translation-profile CUCM

 translate called 8248

dial-peer voice 4001 voip

 description SIP TRUNK TO CUCM

 destination-pattern 8248200

 voice-class codec 1

 session protocol sipv2

 session target ipv4:10.36.4.2

 dtmf-relay rtp-nte

!

dial-peer voice 8248200 pots

 translation-profile incoming CUCM

 incoming called-number 367389

 direct-inward-dial

[Shalid Kurunnan Chalil]

Hi, The disconnect cause indicates that unassigned number. could you please confirm whether you have the number (8248200) defined in the call manager (10.36.4.2). what is SIP trunk inbound call routing configuration in Cisco call manager? ? can you post the screenshot of the trunk configuration? What is the trunk status in Call manager? Regards, Shalid

[Mohammed al Baqari]

Make sure that CUCM sip trunk has the right CSS to route the call. I suggest you use the DNA (https://cucm-ip/dna) to see how the call routed once it reaches cucm. The 404 error is coming from CUCM

[yamikani2g2]

Mohamed okay kind of lost on this  (https://cucm-ip/dna)???

Managed to have calls come in CSS and Route Pattern and Partition did the magic...

thanks

[Rolando Valenzuela]

Can you post the results of debug ccsip events? The reason codes for the failure should be there.

The SIP trunk is already configured on CUCM?

Is the router able to communicate with CUCM?

Are you using the correct interface/IP for your SIP communication?

voice service voip
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

You can find additional information here: https://community.cisco.com/t5/collaboration-voice-and-video/understanding-sip-traces/ta-p/3137704.

Regards.

Rolando A. Valenzuela

[yamikani2g2]

Good day Experts.

1. I think i lack the Inbound route on the CUCM.

2. There is communication between the Call manager and the Cisco Gateway as previously MGCP was configured on it.

3. The number (8248200) is defined in the call manager (10.36.4.2).

4. I really dont know how to check the status of of a SIP trunk on CUCM but it is configured. Its something i have been asking to check. I have been asked to get RTMT to check the SIP calls.

Please advise what the below does the below configuration does? This gateway  will have two SIP trunk..

voice service voip
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

Thats where we are.. now thanks for the URLs I am reading up on them.

[yamikani2g2]

Good day

Mohamed please clarify on this part (https://cucm-ip/dna)??

However what resolve the problem was a route partten that i created the ingress calling worked when i put ! thus allowing all calls coming in.

What i found challenging was Egress calling.

[Shalid Kurunnan Chalil]

Hi,

https://cucm-ip/dna  -->  This is the URL to perform dial number analysis. This allows you to examine/verify your dial plan, 

for an egress call, you can again make use of the dialed number analyzer tool to see if you are able to reach to the gateway. 

Regards,

Shalid.