<div dir="ltr"><div>We're using Asterisk 14.7.6, and we're able to route particular extensions to a corresponding automated member via SIP. Each extension has its own specific content, and all the automated members are configured to handle any of these extensions. We would like to achieve some scaling by putting all the members in a single queue.</div><div><br></div><div>Here are some problems we've run into:</div><div><br></div><div>1) How should these members be declared in queues.conf? We couldn't find much about automated members in the Asterisk Book (

<a href="http://the-asterisk-book.com/1.6/queues.conf.html" target="_blank" style="color:rgb(97,133,198);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(9,9,15)">http://the-asterisk-book.com/1<wbr>.6/queues.conf.html</a>

) nor the wiki (<a href="https://wiki.asterisk.org/wiki/display/AST/Building+Queues">https://wiki.asterisk.org/wiki/display/AST/Building+Queues</a>).</div><div><br></div><div>For example, all of our extensions start with 773, and we're starting with two automated members (declared in sip.conf as CB1 and CB2). </div><div><br></div><div>

<span style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(9,9,15);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">member => SIP/773X@CB1</span>

<br></div><div>

<span style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(9,9,15);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">member => SIP/773X@CB2</span><br></div><div><span style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(9,9,15);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></div><div><span style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(9,9,15);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Is this a valid way to declare these members and the extensions they can handle?</span></div><div>

<div style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration-style:initial;text-decoration-color:initial"><div><div><br class="gmail-Apple-interchange-newline">2) When I configure the queue that way, it looks like calls don't make it to the automated agents. Is there a way to inspect Asterisk to see what extension is being provided to the agents by the queue?<br></div></div></div><div style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration-style:initial;text-decoration-color:initial"><br></div><div style="color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration-style:initial;text-decoration-color:initial"><div>  == Using SIP RTP CoS mark 5</div><div>       > 0x7f32f8013380 -- Strict RTP learning after remote address set to: (IP address of caller):24804</div><div>    -- Executing [77314a31e9c0-6173-4b41-9938-<wbr>be3325139088@default:1] Answer("SIP/1000-00000012", "") in new stack</div><div>       > 0x7f32f8013380 -- Strict RTP switching to RTP target address<span> </span><span style="background-color:rgb(9,9,15);color:rgb(216,209,194);font-family:arial,sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">(IP address of caller)</span>:24804 as source</div><div>    -- Auto fallthrough, channel 'SIP/1000-00000012' status is 'UNKNOWN'</div></div></div><div><br></div><div>Cheers,</div><div>David</div><div><br><div></div></div><br><div><br></div></div>