The Call Routing Test provides verification of System Manager administration for routing a calling party URI to a called party URI.
Use this test to verify that you have administered the system as intended before placing it into service or to get feedback on why a certain type of call is not being routed as expected. No real SIP messages are sent during this test.
This test displays the routing decision process as it uses the SIP routing algorithms. It uses administration from the following forms:
, Dial Patterns, Entity Links, Locations, Policies, Domains, SIP Entities, Time Ranges
After the test has finished, two headings are displayed: Routing Decisions and Routing Decision Process.
The Routing Decisions output contains one line per destination choice (there will be more than one line if there are alternate routing choices; the output will appear in the order that destinations are attempted). Note that each line tells you not only where the INVITE would be routed, but also what the adapted digits and domain would be.
The Routing Decision Process information contains details about how the Routing Decisions were made.
