[Asterisk-Users] VoIP Provider SIP Call Flow

[Andres]

James Rothenberger wrote:

>
> 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.

I know what you mean.  We saw the same issue.  Try version 1.0.3 or 
better.  It should work as you expect it.

>
> Thank you!


-- 
Andres
Network Admin
http://www.telesip.net