Hello!
Please help configure the web sockets in the asterisk so that it can register from SIPML5.
I’am using this document https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5, but i have one problem, my <<res_pjsip_transport_websocket.so>> not running. And I think that’s why I can’t connect via websocket.
Do you have chan_sip loaded as well? Did it load before?
You have autoload enabled so it would automatically load the module if present. Have you checked the console output to see if chan_sip is loaded before?
No. how can I do it? 
Log messages are output to /var/log/asterisk/messages, this is one of the fundamental things that you need to know if you plan on using Asterisk at all. You can also edit sip.conf in /etc/asterisk and set “websocket_enabled = false” in the general section.
If you don’t know what I mean by that you’ll need to read some of the basic Asterisk stuff to get a base understanding. WebRTC is not something you should dive into on Asterisk without that.
I try to config asterisk via FreePBX to support a remote WebRTC client.
Yes, I’m new in this segment, and I want to learn. If you can help me or show the way, please do it.
Send me documentation or links , wich can help me .
You need to pick which way to go: FreePBX or vanilla Asterisk. In the case of FreePBX it configures things and takes care of stuff, so you have to configure it through their interface. Support for that would be on their forums.
Ok .
Thanks a lot .
I’ll ask my question in the FREEPBX forum
