Jonas Kellens
2016-Nov-21 13:05 UTC
[asterisk-users] Asterisk 13.12.2 : strange queue behaviour
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 ? Kind regards. Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161121/5f3b7ea9/attachment.html>
Matthew Jordan
2016-Nov-21 14:17 UTC
[asterisk-users] Asterisk 13.12.2 : strange queue behaviour
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 Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161121/73c9e6b0/attachment.html>
Jonas Kellens
2016-Nov-21 16:05 UTC
[asterisk-users] Asterisk 13.12.2 : strange queue behaviour
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 JordanHello 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. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161121/bab62b2c/attachment.html>