I have 1.8.7.0, Realtime queue table with ringinuse set to 0, callcounter set to yes in sip .conf for my SIP members. Above allows me Queue not sending a call to a member when (s)he is on call(Be it from same Queue or any other call). Member can also transfer(through features.conf) a call without any issue.
call-limit I think is deprecated in 1.8. --Satish Barot Ahmedabad, India On Sat, Jun 22, 2013 at 2:41 PM, Shanavaz E A <[email protected]> wrote: > Hi, > > I use asterisk 1.8. > > My issue is : I have the same SIP members added to two queues. I use > realtime configuration and has set the field ringinuse=0 for both the > queues. But if an extension is answering the call in one queue, and some > new call comes in the second queue, still that extension is ringed. In the > queue_log table I am getting RINGNOANSWER events each second for the > extension until the call gets answered. > > Is this a normal behaviour ? Can we prevent it? Can we set "not to ring" > any queue member if he is answering a call either in the same queue or a > different queue? Pls guide me. > > Regards > Shanavaz. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
