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
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
