i’ve setup webrtc following guide: https://www.asterisk.org/asterisk-15-multi-stream-media-sfu/
test with ‘guestuser’ works fine but when i create user with freepbx with all the same parameters from guestuser but with name 602 (phone number) - in 30seconds after starting video-conference i see error: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel ‘PJSIP/602-000000d8’ for lack of video RTP activity in 30 seconds
i’ve found differences. but i can’t correct them 
added:
[602](+type=endpoint)
webrtc=yes
to pjsip.endpoint_custom.conf
tried also:
[602](+)
type=endpoint
webrtc=yes
reloaded dialplan, restarted asterisk.
but “pjsip show endpoint 602” still shows webrtc=no
why setting doesn’t apply? some syntax problem or… ?