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
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
