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

Joshua Colp jcolp at digium.com
Mon Aug 20 09:53:38 CDT 2012


----- 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
> 
> I suspect I am configuring something wrong, but what is it?

The complete URL to use is http://<asterisk IP address or host>:8088/ws

Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in.

--
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 



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