Hi, I’d like to test and implement audio and video conferencing solutions using freepbx and webrtc. I tried to follow the official asterisk guiu but I can not do much.
You have some advice
Without logs, System Information and your setup information no one is going to guess whats going on.
I have no specific errors, I am looking for a procedure to follow to install webrtc on the freepbx
I have an installation running successfully. 90 % chances of the misconfigured HTTP server. tell me what is your configurations.
Hi,
i have an clean installation of freepbx.
I have registrated a let’s ssl certifcate.
I created two extensions on the pbx and enabled the phone from ucp panell. the phone the interior is called without problems.
I would like to be able to use my own code to create audio and video calls using the current freepbx configuration
is it working now ?
no,
still problems
I got the same problem. I can make a call using the WebRTC Phone in UCP, but not in sipML5.
I got the same results in my browser console as your screenshot. 401 Unauthorized or 403 forbidden. But when I check my Asterisk Log Files, I got the following:
[2019-02-04 02:00:42] NOTICE[4935] chan_sip.c: Registration from ‘“850”<sip:850@mydomain. com>’ failed for ‘XX.XX.XXX.XXX:49960’ - Wrong password
Did you figured it out?
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