[asterisk-users] Asterisk 13.12.2 : strange queue behaviour

[Matthew Jordan]

On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens <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>
> 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
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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