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hi. I am using https://github.com/havfo/WEBRTC-to-SIP to have a web sipcaller. it uses kamailio + rtpengine + web sipcaller (jssip)
right now it is working using internal account and can call to another sip.
But I need to use websipcaller to call to voip provider. I tried to register on my voip provider, and use just outbound websocket proxy, but it not working giving 401
the config I am using is https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
Please help adopt it to relay anything to external sip provider. I don't know what I need, relay, sip trunk or path module.
kamailio configs are very hard for me to understand.
So I need it just as websocket proxy to my voip provider, that doesn't support websocket.
thanks!
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