Unlimited queue timeout

Hi All experts,

I am trying to configure a queue with 3 agents (each one of them is external call which i am trying to execute)

In the background i am playing MOH.

The main target is to have a queue with unlimited loops that trying to call the agents.

In the current configurations what happens is that after 5 tries to call each agent (agent X --> agent Y --> agent Z --> agent X -->agent Y)

Please see queue configurations in the queue.conf and also the way which i call the queue it self



music = 2ndpisheri
strategy = rrordered
timeout = 15
timeoutpriority = conf

member => SIP/xxxxxxxxxxx@abc_1
member => SIP/yyyyyyyyyyy@abc_1
member => SIP/zzzzzzzzzzz@abc_1


exten => _X.,n,Queue(Pishari,5000)

Your feedback is required

Has the call been answered, or is it in early media state? Is it coming from the PSTN?

Your dialplan fragment doesn’t answer the call. PSTN operators generally won’t allow an unanswered call to ring for very long, so, if you can’t get it to an agent quickly, you have to answer it and accept that the caller may get billed for the time they are waiting.

Hi David,

Thanks a lot for your reply.

I am not using strategy of ringall.
I am trying to use rrordered in order to try and each “agent” for 15 sec no more.

The loop which i am trying to do is without end.

Currently i am trying only to test - and in the test - i am not answering - it calls 15 sec trying the second number after another 15 seconds going to the 3rd one.

The call is disconnected when ending to call the 2nd number (for the 2nd time) - meaning i had 5 calls only …

Hope it is more clear

I think you will find the call is cancelled by the caller (by their phone or network), rather than rejected by Asterisk, and you will need to answer before entering Queue to avoid that.

Also, could you confirm that this is not in Germany, as they have special rules about not charging people to wait in queues.

No this is not in Germany.

In my flow i am trying to call to a member which is external number(another operator)
After 15 sec of not receiving a reply i am getting the call back and i am trying to the 2nd agent who is in another operator.

Do you see anything that can be update\change in the configurations of the queue which i have shared?

As @david551 mentioned previously, you’re going to need to actually investigate and identify who is hanging the call up. It is most likely the calling side that is doing so as a result of not answering, which Asterisk would have no control over it.

Failing that you will need to provide the actual console output with SIP traffic (sip set debug on) so it can be seen precisely what is going on.

Hi jcolp,

i opened the SIP debug last packages received are as below:

<— SIP read from UDP:XXX.XXX.XXX.XX:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK3ddd2d13
To: sip:aaaaaa@XXX.XXX.XXX.XX;tag=3809827251-1553766927
From: sip:546400654@yyy.yyy.yyy.yyy;tag=as7b0ae697
Call-ID: 002eed18622c35b64c30051505c91d40@yyy.yyy.yyy.yyy:5060
CSeq: 102 INVITE
Contact: sip:aaaaaa@XXX.XXX.XXX.XX:5060
Content-Length: 0

— (9 headers 0 lines) —
Transmitting (no NAT) to XXX.XXX.XXX.XX:5060:
ACK sip:aaaaaa@XXX.XXX.XXX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK3ddd2d13
Max-Forwards: 70
From: sip:546400654@yyy.yyy.yyy.yyy;tag=as7b0ae697
To: sip:aaaaaa@XXX.XXX.XXX.XX;tag=3809827251-1553766927
Contact: sip:546400654@yyy.yyy.yyy.yyy:5060
Call-ID: 002eed18622c35b64c30051505c91d40@yyy.yyy.yyy.yyy:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘002eed18622c35b64c30051505c91d40@yyy.yyy.yyy.yyy:5060’ in 32000 ms (Method: INVITE)
== Spawn extension (TestMoh, 0797406003, 8) exited non-zero on ‘SIP/CELLACT_GB_1-00a376a5’
Really destroying SIP dialog ‘13981547-3809827170-2087904627@SBC1.aaaaa.bb.cc’ Method: ACK

Please advise if any packages before are also required?

You need to provide the FULL SIP trace. Providing only a portion doesn’t show the entire picture, because what you’ve provided just means Asterisk sent a CANCEL and cancelled an outgoing call attempt.

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