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<div class="moz-cite-prefix">On 21-11-16 17:20, Matthew Jordan
wrote:<br>
</div>
<blockquote
cite="mid:CAN2PU+6Qtw-FhU9AMuRfPkO6YYDNeTjt0XqXisPZTtWiQzvUxw@mail.gmail.com"
type="cite">
<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 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: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
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: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 moz-do-not-send="true"
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>
</div>
</div>
</div>
</blockquote>
<br>
Hello<br>
<br>
indeed no debug log output. Therefore I need to know what to filter
because there is a lot of information written.<br>
<br>
"you have not configured the Queue to ring the Queue Member while
they are busy"<br>
<br>
--> where would I configure this ?<br>
<br>
I have in my realtime MySQL tables 'queues' a column 'ringinuse'
with value 'no'.<br>
<br>
<br>
I would expect that the call does enter the call queue but when the
member is called there is a 'busy' notification for that member.
This way the dialplan can continue with the next step.<br>
<br>
Now the call 'hangs' at the queue application until this queue()
command can continue.<br>
<br>
Is this normal behaviour in version 13.12.2 ? Personally I prefer
the previous behaviour of the Queue application.<br>
<br>
<br>
<br>
Kind regards.<br>
<br>
J.<br>
<br>
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