Thanks again!
all my extensions have insecure=invite,port. I think Asterisk is forwarding
the call as: from anonymous.
right now when the user makes a selection I use a GOTO in Asterisk to send
it to an Asterisk CONTEXT and from that CONTEXT I use:
Dial(SIP/opensips/1005) opensips is the pe
Thank for the input!!
When I do, it fails and a SIP trace shows error: Failed to authenticate on
INVITE to "Anonymous"
I'm using basically using the "residential" script with some hooks for
Asterisk. Asterisk is doing some IVR, once a department / individual is
selected I want asterisk t
