[asterisk-users] Pjsip and Call limit
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
More information about the asterisk-users