[SR-Users] Help with SIP over Websocket audio call: 405 Method Not Allowed

[Brad Johns]

Hi,

New to Kamailio.  I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
 They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.

However, after registration, the users can't place an audio call.  I see no
ringing on the remote browser.  Can anyone help with clues or debug?  In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE.  However, nothing
indicating a call.

I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting
an error response to browser UA of "405:  Method Not Allowed".  I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:

        $avp(oexten) = $rU;
        if (!lookup("location")) {
                $var(rc) = $rc;
                route(TOVOICEMAIL);
                t_newtran();
                switch ($var(rc)) {
                        case -1:
                        case -3:
                                send_reply("404", "Not Found");
                                exit;
                        case -2:
                                send_reply("405", "TEST:  Method Not
Allowed");
                                exit;
                }
        }


The switch case is returning -2, for some reason.

Any help in debugging this appreciated.
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