Hi, can you share with us the asterisk dialplan part where you call the
Dial() application?
On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve <swwilkins(a)mitre.org> wrote:
Hello All,
I am looking for a Diagram or such that shows the flow of SIP traffic for
a WebRTC Client1 => WebRTC Client2 call using Kamailio in front of
Asterisk.
I am unable to get Asterisk to find the correct registered clients, which
are registered in Kamailio and am hoping verifying the flow will help give
me a clue as to what is going on. E.g. Using chrome and tryit-pjsip I have
Client1, and Client2 registered in Kamailio. However when I try to connect
Client1 to Client2 (make a call), Asterisk has no clue where Client1 and
Cleint2 are registered to.
Thank you!
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