[asterisk-users] Pjsip and Call limit

Administrator TOOTAI admin at tootai.net
Thu Dec 27 13:14:05 CST 2018


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


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