Good day.

First off let me explain the scenario I'm testing.

This isn't a pentest per say that I'm trying to do more of a stress test. We are currently migrating our enterprise PABX from a 15 year old Philips PABX to a Cisco CUCM. This obviously means a switch from TDM to SIP/RTP. Our environment consists of a 800 seat call center and back office phones/faxes numbering around 3500.

The network is vast and complex with 8 core switches with 10gig fiber links between the core and the distribution switches (6500 blades) spread across 4 buildings geographically co located.

There is an obvious concern over introducing SIP/RTP traffic to an already busy LAN. The specific area we want to test is the traffic between the call center application's SIP proxy and the CUCM. We have a developement and testing system whre I am trying to test the SIP proxy with Sipp but cannot get thecall flow complete. The registration process works fine, I have installed and enabled Openssl and the required dependancies so all good there. The SIP error I keep getting is '604 does not exist anywhere' while expecting a '180'. Wireshark traces confirm. I've tried the built in UAS and UAC scenarios but cannot get the call flow to a '200 OK'

Is there anyone here with better working knowledge of sipp than I have been able to find in a google search?

Thanks in advance