Re: [asterisk-users] Pjsip and Call limit
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: Le 27/12/2018 à 20:42, Social Boh a écrit : Hello, you have to use GROUP and GROUP_COUNT functions. Well, could be done for extensions but for queue ? Does it mean ringinuse is useless ? [...] El 27/12/2018 a las 14:14, Administrator TOOTAI escribió: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pjsip and Call limit
Le 27/12/2018 à 20:42, Social Boh a écrit : Hello, you have to use GROUP and GROUP_COUNT functions. Well, could be done for extensions but for queue ? Does it mean ringinuse is useless ? [...] El 27/12/2018 a las 14:14, Administrator TOOTAI escribió: 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pjsip and Call limit
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail: MWI problem / pjsip (13.24.0)
On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote: > Hi! > > I just want to say, that 13.24.1 doesn't fix the problem described in > the posts above. You're going to need to file an issue[1] with traces and actual configuration. [1] https://issues.asterisk.org/jira -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail: MWI problem / pjsip (13.24.0)
Hi! I just want to say, that 13.24.1 doesn't fix the problem described in the posts above. Regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
