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G3261
Participant

Understanding Tracing a call

Hello All Experts, I have been trying to figure this out but not sure how to? If I want to trace calls  INBOUND, OUTBOUND, SIP, Fax calls etc. so I can figure out the PATH and gateways/trunk involve in  a call from a site to PSTN and PSTN to site. What is the best way of figuring this out? Is Cisco's DNA or RTMT tool can do that? Is there any other way I can achieve this? Thank you

8 REPLIES 8
Kaloyan
Cisco Employee

Hello,

 

If enabled, you can utilize the Call Detail Records (CDRs) search https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/11_5_1/car/CUCM_BK_C72B9385_00_cdr-analysis-and-reporting_1151/CUCM_BK_C72B9385_00_cdr-analysis-and-reporting_1151_chapter_011010.html. The originating and destination device names and IP addresses are included. 

 

Perhaps the most user-friendly way will be to use RTMT and the Real Time Data https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html.

mdrangell22
Beginner

Hello if you want to trace some SIP calls from PSTN to your office or viceversa and you are using a CISCO SIP GW/CUBE you can enable some debugs here.

 

The most common SIP debug on GW/CUBE is debug ccsip messages where you can see the SIP messages send and received that form a call. I have to say It very useful

 

If you want to see the traces from SIP Trunk  created in your CUCM connected to any other company or branches you have to do it using RTMT with 2 option

 

First One: Once RTMT is opened you have to choose Trace and Logging central --> Collect Files --> Cisco Call Manager--> Choose the correct CUCM node --> After that you have to choose the time where you want to collect the logs and where you want to save those files (Ex=Desktop) ...... This is not the best way because you will see a lot of traces calls and it could be a little confuse......

 

Second One: Once RTMT is opened you will see at the botton left the section voice/video option and after that you should choose  SIP Activity where you can see some options to complete as Calling/Called number and the respective date and RTMT will show you the calls with the information you provided... You have to choose one and you will see the traces call flow.... This is the best option.

 

In both cases you have to enable the traces and I think this is over CUCM SERVICEABILITY Page

Thank you so much for details. However, my problem still continues because in order to troubleshoot a SIP call issue, I believe it is critical I understand call flow first....

I have to be honest I haven't had that error before... I have had 401 Error that is referred to error authentication.

 

I Think that you are receiving that error because the other end doesn't know or doesn't found the number you are sending over the SIP Trunk....

fair enough....I do appreciate your input. I will continue to work on this and update this conversation if I have a solution.

well..it turned out to be a CSS/PTN issue for outbound calls. It is fixed now.

Scott Leport
Participant

Hey Guys,

 

Worth checking out TranslatorX if you want to look at SIP traces quickly. It will even generate you a ladder diagram too if you're into that sort of thing!

https://translatorx.org/

 

As for your issue, it looks like CUCM can't find the number so I'd check the configuration on CUCM for this number.

Yes...I have been using Translator X.

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