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

Andrew Latham lathama at gmail.com
Mon Aug 20 14:00:25 CDT 2012


On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro <jcastro at instant.com.br> wrote:
> 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
> http://www.instant.com.br
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>
> --

Juan

Matt just opened
https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document
some of this.  Feel free to pipe in.

-- 
~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~



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