[Asterisk-Users] VoIP Provider SIP Call Flow

[James Rothenberger]

I am testing a call flow in which an inbound SIP call (to the Asterisk from 
a PSTN connection from a SIP VoIP provider) is not answered (nobody there 
and no voicemail) and the call is terminated on the PSTN side.  After the 
SIP CANCEL is sent to the Asterisk from the PSTN, The SIP phone sends a 487 
response back to the Asterisk (Request Terminated) as it should.  What is 
NOT occurring is that the 487 is NOT propagated back to the provider.  The 
asterisk simply sends an OK back in acknowledgment of the initial CANCEL. 
How do I force the Asterisk to send the 487?  I also have the same signaling 
problem with 486, 481, and 408 SIP responses.  I am using asterisk v1.0.0.

Thank you!
-- 
James Rothenberger