Queue not time up when no agents answer

Asterisk Asterisk Support
1

OK, I have a very simple queue config and I have two incoming connections, one is through a newrock gateway and the other is a sip trunk.
both go through same configuration for handling incoming calls and on extension 105, both try to use my test-queue. Everything works as it should, on the sip trunk. But when I call the gateway numbers, the call goes through the queue, but never stops/timed-out/reject even if the caller hangup!
extension.conf:
[fxo-hasin]
exten => s,1,NoOp(${CALLERID(num)})
exten => s,n,Answer
exten => s,n,Background(8)
exten => s,n,WaitExten(10)
exten => s,n,Hangup
exten => 105,1,Queue(test-queue)

queue.conf:
[test-queue]
strategy=linear
timeout=15
retry=1
member=>SIP/80105

sample result:
== CDR updated on SIP/fxo-00000715
– Executing [105@fxo-hasin:1] Queue(“SIP/fxo-00000715”, “test-queue”) in new stack
– Started music on hold, class ‘default’, on SIP/fxo-00000715
== Using SIP RTP CoS mark 5
– SIP/80105-00000718 is ringing
– SIP/80105-00000718 is ringing
– SIP/80105-00000718 is ringing
----------------------------- I disconnected the call from the caller point here -----------------------------
– SIP/80105-00000718 is ringing
– SIP/80105-00000718 is ringing
– SIP/80105-00000718 is ringing
– SIP/80105-00000718 is ringing
– Nobody picked up in 15000 ms
– Stopped music on hold on SIP/192.168.240.250-0000072a
== Using SIP RTP CoS mark 5
– SIP/80105-00000719 is ringing
– SIP/80105-00000719 is ringing
– SIP/80105-00000719 is busy
– SIP/80105-00000719 is ringing
– SIP/80105-00000719 is ringing
…
To stop the 80105 extension from ringing, I should pick it up.

2

Please provide SIP debugging proving that the peer actually sent a BYE. I imagine that there is either no disconnect supervision on the analogue line, or the gateway is not configured to detect it.

If you want a time out on the whole queue operation, you need to specify it on the Queue application call.

3

OK, unfortunately I couldn’t get the SIP debug information. As the system is under heavy load already and the sip debug data is huge.

4

hey , I have some kind of same problem with newrock, as the call enter the queue the bussy tone dont send to callers if the queue members are on call.
and if you configured the newrock and all other things are proprly functioning , I would be thankfull if you suggest me with your own configuration between the asterisk and newrock,
با تشکر:wink: