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

[James Mortensen]

Hi Sven,

I tried out your changes. I had to replace the $_SERVER['REMOTE_ADDR'] with
Java's request.getRemoteAddr() since I'm using Jetty not Apache.  I got the
same results you got, which I also get using the something.invalid header.
The peer connects from Chrome, I can dial my cellphone and make it ring,
but the Chrome sipml5 client drops the call when the phone starts ringing.
When I answer, the cellphone stays connected, but there is no audio.

My suggestion is to post your changes to the user interface on the doubango
Google Group as it will mean people don't need to modify the code to
connect to Asterisk WS.
https://groups.google.com/forum/?fromgroups=#!forum/doubango.  See if they
can incorporate your changes so we don't have to modify the library after
each update.

As far as the IP address goes, I'm not sure what this is doing since I
still see the invalid domain in my SIP traces.

James



>*I did some changes to the sipml5 client and wanted to share this with you

guys... Actually only 2 simple changes...*https://github.com/mailsvb/sipml5

*- The main config section has been splitted and made a little more
flexible, see *http://i45.tinypic.com/10x59o7.png
- Main call.html file has been renamed to .php and some code has been added
that will replace the "something.invalid" with the actual IP of your client
PC.

Currently I am able to register and at least make my softphone ring ;-) As
soon as I answer the outgoing call from sipml5 in the softclient, I get an
error in sipml5...

You can find my console output here http://pastebin.com/jdkXSMSD
I will continue investigating tomorrow...

best regards,
Sven


-- 
James Mortensen
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