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

[Juan Castro]

Hoo-hah. It registers. Progress!

Now... media. Or not.

On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp <jcolp at digium.com> wrote:
> ----- Original Message -----
>> >
>> > 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.
>>
>> Well, I did the following changes in sipml5 and now I get a "Bad
>> Request" on REGISTER, instead of 404. Clearly, I'm still missing
>> something. Here are the changes I made:
>
> You are probably getting hit by a bug in Asterisk 11 that has been fixed.
>
> It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration.
>
> --
> 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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-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
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