Juan Castro
2012-Aug-17 18:45 UTC
[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100
Andrew Latham
2012-Aug-17 19:54 UTC
[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro <jcastro at instant.com.br> wrote:> I see no indication of how to do this in sip.conf, and when I start > Asterisk, it doesn't wait on port 80. > > Greetings, > > -- > Juan Carlos Castro y Castro > Instant Solutions - Telefonia Gerando Resultado > http://www.instant.com.br > Principais capitais: 4063-6100 > Demais regi?es: (11)4063-6100 > > --Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. --- manager.conf --- [general] enabled = yes webenabled = yes --- http.conf --- [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 -- ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~
James Mortensen
2012-Aug-22 16:32 UTC
[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
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 youguys... 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120822/d1dbb58b/attachment.htm>