[sr-dev] [kamailio/kamailio] relay to voip provider / sip trunk / path? (Issue #3990)

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! -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3990 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3990(a)github.com>