Pjsip And Call Limit

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Asterisk Users 2 Comments

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

2 thoughts on - Pjsip And Call Limit

  • Le 27/12/2018 à 20:42, Social Boh a écrit :

    Well, could be done for extensions but for queue ? Does it mean ringinuse is useless ?

    Daniel

  • Hello,

    ringinuse still working on queue configuration:

    ringinuse=no

    Ring the extension only if not in use.

    Regards


    I’m SoCIaL, MayBe

    El 27/12/2018 a las 17:11, Administrator TOOTAI escribió: