[asterisk-users] Pjsip and Call limit

[Administrator TOOTAI]

Hello,

I'm used to set call-limit in sip.conf Now I switched one customer 
Asterisk to 16 version and can't get the behavior back, as well for 
extensions as for queues.

I set ringinuse=no for queues and have max_audio_streams = 1 
max_video_streams = 0. I wanted to add max_calls = 1 but this parameter 
is not accepted.

Thanks for any hint

-- 
Daniel