<html><body><div style="color:#000; background-color:#fff; font-family:arial, helvetica, sans-serif;font-size:10pt"><div><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><span>Hi,</span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><span>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.</span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family:
arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><span>Any other methods ?</span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><br><span></span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><span>Thanks & Regards</span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style: normal;"><span>Shanavaz.</span></div><div style="color: rgb(0, 0, 0); font-size: 13.3333px; font-family: arial,helvetica,sans-serif; background-color: transparent; font-style:
normal;"><span></span></div><div><br></div> <div style="font-family: arial, helvetica, sans-serif; font-size: 10pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Shanavaz E A <shanavazea@yahoo.com><br> <b><span style="font-weight: bold;">To:</span></b> "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com> <br> <b><span style="font-weight: bold;">Sent:</span></b> Saturday, June 22, 2013 1:11 PM<br> <b><span style="font-weight: bold;">Subject:</span></b> [asterisk-users] Queue Ring inuse is shared ?<br> </font> </div> <div class="y_msg_container"><br><div id="yiv4859833960"><div><div style="color:#000;background-color:#fff;font-family:arial, helvetica, sans-serif;font-size:10pt;"><div>Hi,</div><div><br></div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial,
helvetica, sans-serif;background-color:transparent;font-style:normal;">I use asterisk 1.8.</div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;"><br></div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;"> 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.</div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;"><br></div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;">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.</div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;"><br></div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;">Regards</div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica,
sans-serif;background-color:transparent;font-style:normal;">Shanavaz.</div><div style="color:rgb(0, 0, 0);font-size:13.3333px;font-family:arial, helvetica, sans-serif;background-color:transparent;font-style:normal;"><br></div></div></div></div><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com </a>--<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br></div> </div> </div> </div></body></html>