[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Joshua Colp]

----- Original Message -----
> On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro <jcastro at instant.com.br>
> wrote:
> > On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
> > <jcastro at instant.com.br> wrote:
> >> I still get "unauthorized" from sipml5 with these modifications. I
> >> used port 80 instead of 8088 (no other webserver listening on 80),
> >> was
> >> that wrong?
> >
> > Correction. It's actually "Failed to connect to the server". I set
> > the
> > proxy address and port correctly in sipml5's call.htm (it registers
> > on
> > Kamailio).
> 
> ...which is in fact a 404 response from Asterisk. Here's the response
> I received: http://users.vialink.com.br/jcastro/refused.cap

Well, of course unless you changed the port as you did in which case 80 in the URL instead of 8088. That is all!

As I've said previously though, you won't get bidirectional audio or video flowing so trying that will fail, and it's known that it will fail.

--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org