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SIP Call Tracing

We're running CUCM 6.1(1). We've also invested in a NICE recording solution. We've followed their document to configure the Communications Manager. We've set the desired line presences up for Automatic Call Recording Enabled. We've defined and verified the SIP trunk configuration, the Recording Profile, End User and Application User configurations, etc.

NICE has indicated that everything “looks” correct. But they state they never see any RTP packets. Well, a wireshark capture shows that there is NO SIP traffic (so they're not going to see any RTP). There are no ACLs or firewalls between the phone VLAN and the Voice Server VLAN (and the NICE servers and the CallManager Publisher sit on the same VLAN on the same switch).

I think its time to do a serviceability trace but I'm unsure how to set it up. I'm assuming that I would navigate to Trace Configuration > Server [select the Publisher] > Service Group [select CM Services] > Service [Cisco CallManager].

So the two obvious trace fields might be “Enable SIP Stack Trace” and “Enable SIP Call Processing Trace” Since we're doing call recording, I like to see either a BIB (Built In Bridge) trace or Call Recording trace (no such luck).

What other Trace Fields might be useful to see why CallManager or the specific phone is not kicking off any SIP traffic (no INVITE or any other SIP traffic)?


Re: SIP Call Tracing

To answer your first question about collecting traces:

That's the best guide I know of.

For tracing SIP through a trace, I suggest triple combo. Here's a link:

You can track a SIP call by the mandatory SIP header "Call-ID". This will remain the same throughout a call.

You should be aware that CUCM will only allow SIP calls from IP addresses that it has SIP trunks defined for. If you have destination IP 'a' and the device sources SIP from IP 'b', you will not take the call.



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