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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130622/69231c50/attachment.htm>
Hi, I found in another mail that setting call-limit=1 in the sip configuration works. I tried that. It works but in that case the agents are not able to transfer the call to another extension, because only one call is allowed at a time. Any other methods ? Thanks & Regards Shanavaz. ________________________________ From: Shanavaz E A <shanavazea at yahoo.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: Saturday, June 22, 2013 1:11 PM Subject: [asterisk-users] Queue Ring inuse is shared ? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130623/accb7b27/attachment.htm>
On 22 June 2013 10:11, Shanavaz E A <shanavazea at yahoo.com> 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. >Should that not be "ringinuse = no"? -Barry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130624/844ea943/attachment.htm>
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 <shanavazea at yahoo.com> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130625/bdaa5160/attachment.htm>