SIP Persistence - Debugging

I have a couple of asterisk servers sitting as pool members that are tied to a vip. The vip is configured for round robin load sharing. Both asterisk servers no only register sip clients but also, proxy ip calls in and out. Call flow example is as follows:

 

 

ata-->asterisk-->soft switch-->tdm

 

 

(ata=analog telephone adapter)

 

(tdm=time division multiplexing)

 

 

I have created a sip persistence profile with the persistence tag of call-id. How can I effectively debug the persistence profile that I have created to ensure that the call-id tag is working properly?

 

 

Thanks in advance.