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On 21-11-16 15:17, Matthew Jordan wrote:<br>
<blockquote
cite="mid:CAN2PU+7VRX7JZWK7Dds8S+WC+HHud_ATSW7QdvhjKdH9o1OUbQ@mail.gmail.com"
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 moz-do-not-send="true"
href="mailto:jonas.kellens@telenet.be" target="_blank">jonas.kellens@telenet.be</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <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@<wbr>CallFromQueue-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@<wbr>CallFromQueue-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@<wbr>CallFromQueue-00000040;2",
"") in new stack<br>
[Nov 21 08:20:45] pbx.c: Executing
[mysip692@CallFromQueue:3] Dial("Local/mysip692@<wbr>CallFromQueue-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_signature" data-smartmail="gmail_signature">Matthew
Jordan<br>
</div>
</div>
</div>
</blockquote>
<br>
<br>
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@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@CallFromQueue-0000081a;2", "exten = mysip692")
in new stack<br>
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@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-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-f0d9ab0102a7><br>
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-00004e6a left
'native_rtp' basic-bridge
<fed056d3-669a-493d-a4bd-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>
<br>
Kind regards.<br>
<br>
J.<br>
<br>
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