<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens <span dir="ltr"><<a href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
  
    
  
  <div bgcolor="#FFFFFF"><span class="gmail-">
    On 21-11-16 15:17, Matthew Jordan wrote:<br>
    <blockquote type="cite">
      <div dir="ltr">
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Mon, Nov 21, 2016 at 7:05 AM,
            Jonas Kellens <span dir="ltr"><<a href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>></span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
              <div bgcolor="#FFFFFF"> <font face="Helvetica, Arial, sans-serif">Hello<br>
                  <br>
                  when using Asterisk version 13.12.2 I notice that it
                  takes up to 30 seconds (sometimes even longer) for a
                  call queue to call its members.<br>
                  <br>
                  Example 1 :<br>
                  <br>
                  [Nov 21 08:17:57] pbx.c: Executing
                  [queue@pbx-routing:15] Queue("SIP/incoming-00000246",
                  "myqueue1,,,,300,,,") in new stack<br>
                  [Nov 21 08:17:57] res_musiconhold.c: Started music on
                  hold, class 'default', on channel
                  'SIP/incoming-00000246'<br>
                  <br>
                  [Nov 21 08:18:26] pbx.c: Executing
                  [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQ<wbr>ueue-0000003c;2",
                  "") in new stack<br>
                  [Nov 21 08:18:26] app_queue.c: Called
                  Local/mysip692@CallFromQueue<br>
                  [Nov 21 08:18:26] pbx.c: Executing
                  [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQ<wbr>ueue-0000003c;2",
                  "SIP/mysip692") in new stack<br>
                  [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692<br>
                  <br>
                  <br>
                  Example 2 :<br>
                  <br>
                  [Nov 21 08:20:11] pbx.c: Executing
                  [queue@pbx-routing:15] Queue("SIP/incoming-00000255",
                  "myqueue1,,,,300,,,") in new stack<br>
                  [Nov 21 08:20:11] res_musiconhold.c: Started music on
                  hold, class 'default', on channel
                  'SIP/incoming-00000255'<br>
                  <br>
                  [Nov 21 08:20:45] app_queue.c: Called
                  Local/mysip692@CallFromQueue<br>
                  [Nov 21 08:20:45] pbx.c: Executing
                  [mysip692@CallFromQueue:1] NoOp("Local/mysip692@CallFromQ<wbr>ueue-00000040;2",
                  "") in new stack<br>
                  [Nov 21 08:20:45] pbx.c: Executing
                  [mysip692@CallFromQueue:3] Dial("Local/mysip692@CallFromQ<wbr>ueue-00000040;2",
                  "SIP/mysip692") in new stack<br>
                  [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692<br>
                  <br>
                  <br>
                  I did not see this behaviour in previous Asterisk
                  versions.<br>
                  <br>
                  Could this be a bug ?<br>
                  <br>
                </font></div>
            </blockquote>
            <div><br>
            </div>
            <div>There's not enough information here to know what is
              preventing the call from occurring.</div>
            <div><br>
            </div>
            <div>I'd look at a debug log between the caller entering the
              Queue and the outbound call being made. That should
              illustrate what is causing the delay. </div>
          </div>
          <div><br>
          </div>
          -- <br>
          <div class="gmail-m_-2721457819232755237gmail_signature">Matthew
            Jordan<br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    <br></span>
    Hello<br>
    <br>
    <br>
    and what exactly am I looking for in the debug logs ?<br>
    <br>
    I have generated debug output and re-produced the issue.<br>
    <br>
    <br>
    Again 23 seconds before calling the queue member :<br>
    <br>
    [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15]
    Queue("SIP/incoming-00004e6e", "myqueue1,,,,300,,,") in new stack<br>
    [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
    'default', on channel 'SIP/incoming-00004e6e'<br>
    <br>
    [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1]
    NoOp("Local/mysip692@<wbr>CallFromQueue-0000081a;2", "") in new stack<br>
    [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue<br>
    [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2]
    NoOp("Local/mysip692@<wbr>CallFromQueue-0000081a;2", "exten = mysip692")
    in new stack<br>
    [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
    Dial("Local/mysip692@<wbr>CallFromQueue-0000081a;2", "SIP/mysip692") in
    new stack<br>
    [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692<br>
    [Nov 21 16:23:56] app_dial.c: SIP/mysip692-00004e86 is ringing<br>
    [Nov 21 16:23:56] app_queue.c:
    Local/mysip692@CallFromQueue-<wbr>0000081a;1 is ringing<br>
    <br>
    <br>
    <br>
    Could it be that it is because my Queue member 'mysip692' is
    occupied in another bridge (call) ?<br>
    <br>
    This I see in the logs just before the Call Queue starts calling the
    queue member :<br>
    <br>
    [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
    'SIP/mysip-00004e6a' and 'SIP/incoming-00004e63' in stack<br>
    [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-00004e63
    left 'native_rtp' basic-bridge
    <fed056d3-669a-493d-a4bd-<wbr>f0d9ab0102a7><br>
    [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-00004e6a left
    'native_rtp' basic-bridge
    <fed056d3-669a-493d-a4bd-<wbr>f0d9ab0102a7><br>
    <br>
    <br>
    A bit too coincidal, no ?<br>
    <br>
    So then it has something to do with the bridging ?<br>
    <br>
    <br>
    <br>
    I did not have this behaviour in previous Asterisk versions.<br><br></div></blockquote><div><br></div><div>Those aren't debug logs. Instructions for generating debug information can be found on the wiki:</div><div><br></div><div><a href="https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information">https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information</a></div><div><br></div><div>That being said, if the Queue Member is currently busy (which will be denoted by their device state), and you have not configured the Queue to ring the Queue Member while they are busy, then I would expect any new caller to hang out in the Queue until that Member is available.</div></div><div><br></div>-- <br><div class="gmail_signature">Matthew Jordan<br>Digium, Inc. | CTO<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div>
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