[asterisk-users] Asterisk 13.12.2 : strange queue behaviour
Jonas Kellens
jonas.kellens at telenet.be
Mon Nov 21 12:14:50 CST 2016
On 21-11-16 17:20, Matthew Jordan wrote:
>
>
> On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> On 21-11-16 15:17, Matthew Jordan wrote:
>>
>> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
>> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>>
>> Hello
>>
>> 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.
>>
>> Example 1 :
>>
>> [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15]
>> Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack
>> [Nov 21 08:17:57] res_musiconhold.c: Started music on hold,
>> class 'default', on channel 'SIP/incoming-00000246'
>>
>> [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:1]
>> NoOp("Local/mysip692 at CallFromQueue-0000003c;2", "") in new stack
>> [Nov 21 08:18:26] app_queue.c: Called
>> Local/mysip692 at CallFromQueue
>> [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:3]
>> Dial("Local/mysip692 at CallFromQueue-0000003c;2",
>> "SIP/mysip692") in new stack
>> [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692
>>
>>
>> Example 2 :
>>
>> [Nov 21 08:20:11] pbx.c: Executing [queue at pbx-routing:15]
>> Queue("SIP/incoming-00000255", "myqueue1,,,,300,,,") in new stack
>> [Nov 21 08:20:11] res_musiconhold.c: Started music on hold,
>> class 'default', on channel 'SIP/incoming-00000255'
>>
>> [Nov 21 08:20:45] app_queue.c: Called
>> Local/mysip692 at CallFromQueue
>> [Nov 21 08:20:45] pbx.c: Executing [mysip692 at CallFromQueue:1]
>> NoOp("Local/mysip692 at CallFromQueue-00000040;2", "") in new stack
>> [Nov 21 08:20:45] pbx.c: Executing [mysip692 at CallFromQueue:3]
>> Dial("Local/mysip692 at CallFromQueue-00000040;2",
>> "SIP/mysip692") in new stack
>> [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692
>>
>>
>> I did not see this behaviour in previous Asterisk versions.
>>
>> Could this be a bug ?
>>
>>
>> There's not enough information here to know what is preventing
>> the call from occurring.
>>
>> 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.
>>
>> --
>> Matthew Jordan
>
>
> Hello
>
>
> and what exactly am I looking for in the debug logs ?
>
> I have generated debug output and re-produced the issue.
>
>
> Again 23 seconds before calling the queue member :
>
> [Nov 21 16:23:33] pbx.c: Executing [queue at pbx-routing:15]
> Queue("SIP/incoming-00004e6e", "myqueue1,,,,300,,,") in new stack
> [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
> 'default', on channel 'SIP/incoming-00004e6e'
>
> [Nov 21 16:23:56] pbx.c: Executing [mysip692 at CallFromQueue:1]
> NoOp("Local/mysip692 at CallFromQueue-0000081a;2", "") in new stack
> [Nov 21 16:23:56] app_queue.c: Called Local/mysip692 at CallFromQueue
> [Nov 21 16:23:56] pbx.c: Executing [mysip692 at CallFromQueue:2]
> NoOp("Local/mysip692 at CallFromQueue-0000081a;2", "exten =
> mysip692") in new stack
> [Nov 21 16:23:56] pbx.c: Executing [mysip692 at CallFromQueue:3]
> Dial("Local/mysip692 at CallFromQueue-0000081a;2", "SIP/mysip692") in
> new stack
> [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
> [Nov 21 16:23:56] app_dial.c: SIP/mysip692-00004e86 is ringing
> [Nov 21 16:23:56] app_queue.c:
> Local/mysip692 at CallFromQueue-0000081a;1 is ringing
>
>
>
> Could it be that it is because my Queue member 'mysip692' is
> occupied in another bridge (call) ?
>
> This I see in the logs just before the Call Queue starts calling
> the queue member :
>
> [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
> 'SIP/mysip-00004e6a' and 'SIP/incoming-00004e63' in stack
> [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-00004e63
> left 'native_rtp' basic-bridge <fed056d3-669a-493d-a4bd-f0d9ab0102a7>
> [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-00004e6a
> left 'native_rtp' basic-bridge <fed056d3-669a-493d-a4bd-f0d9ab0102a7>
>
>
> A bit too coincidal, no ?
>
> So then it has something to do with the bridging ?
>
>
>
> I did not have this behaviour in previous Asterisk versions.
>
>
> Those aren't debug logs. Instructions for generating debug information
> can be found on the wiki:
>
> https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>
> 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.
>
> --
> Matthew Jordan
Hello
indeed no debug log output. Therefore I need to know what to filter
because there is a lot of information written.
"you have not configured the Queue to ring the Queue Member while they
are busy"
--> where would I configure this ?
I have in my realtime MySQL tables 'queues' a column 'ringinuse' with
value 'no'.
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.
Now the call 'hangs' at the queue application until this queue() command
can continue.
Is this normal behaviour in version 13.12.2 ? Personally I prefer the
previous behaviour of the Queue application.
Kind regards.
J.
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