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Summary:ASTERISK-16277: eventwhencalled=yes does not generate any events
Reporter:Claudio Villalobos (devmod)Labels:
Date Opened:2010-06-21 21:20:27Date Closed:2011-06-07 14:05:08
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_queue
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) issue_17541_ami_capture.txt
( 1) issue_17541_full_log.txt
( 2) manager.conf
Description:Asterisk SVN-branch-1.6.2-r265699M

I believe I am seeing the same issue describe on this ticket on the 1.6.2 branch:

https://issues.asterisk.org/view.php?id=11385

I am not seeing any events generated when an agent is being called.
I tested with eventwhencalled=yes and eventwhencalled=vars and nothing is going out.
Comments:By: Paul Belanger (pabelanger) 2010-06-22 07:06:55

We need to see your .config file and debug log (see below), be sure to capture the manager events in the debug log.

---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Claudio Villalobos (devmod) 2010-06-22 12:19:37

Updated my 1.6.2 branch to the latest - 271555 - and still observe the issue.
I attached the logs I obtained on this updated installation.

--- FYI this is what I have in manager.conf ---

read = system,call,agent,user,dialplan
By: Tilghman Lesher (tilghman) 2010-06-29 12:01:09

Could you upload your complete manager.conf (with any passwords XXX'ed out)?
By: Claudio Villalobos (devmod) 2010-06-29 12:18:34

Attached it to the ticket.
By: Tilghman Lesher (tilghman) 2010-06-29 17:05:25

Okay, please upload your queues.conf, too.  I just tried reproducing this, and I'm getting the AgentCalled event fine (which is one of the eventwhencalled events):

Event: AgentCalled
Privilege: agent,all
Queue: atendimento
AgentCalled: SIP/vidphone
AgentName: SIP/vidphone
ChannelCalling: SIP/103-00000000
DestinationChannel: SIP/vidphone-00000001
CallerIDNum: 103
CallerIDName: Tilghman Lesher
Context: digium
Extension: 8168
Priority: 2
Uniqueid: 1277848959.0

Console:
   -- Executing [8168@digium:1] Answer("SIP/103-00000000", "") in new stack
   -- Executing [8168@digium:2] Queue("SIP/103-00000000", "atendimento,tk,,,900") in new stack
   -- Started music on hold, class 'default', on channel 'SIP/103-00000000'
 == Using SIP RTP CoS mark 5
 == Using SIP VRTP CoS mark 6
   -- Called SIP/vidphone
   -- SIP/vidphone-00000001 is ringing
   -- Stopped music on hold on SIP/103-00000000
 == Spawn extension (digium, 8168, 2) exited non-zero on 'SIP/103-00000000'

queues.conf:
[atendimento]
musicclass = default
context = queuereturn
strategy = fewestcalls
wrapuptime=0
announce-frequency = 165
queue-thankyou = /var/lib/asterisk/sounds/en/tt-somethingwrong
monitor-format = gsm
joinempty=yes
member => SIP/vidphone
eventwhencalled=yes


By: Claudio Villalobos (devmod) 2010-06-29 18:08:16

Here it is.

 [general]
 persistentmembers = no
 autofill = yes
 monitor-type = MixMonitor
 shared_lastcall=no
 setinterfacevar=no
 setqueueentryvar=no
 setqueuevar=no
 announce-position-limit = 5
 eventwhencalled=yes
 
 [callcenter]
 musicclass = default
 strategy = rrmemory
 context = callcenter
 timeout = 30
 retry = 2
 wrapuptime=5
 announce-frequency = 0
 periodic-announce-frequency=30
 announce-holdtime = no
 joinempty=yes
 ringinuse=no
 periodic-announce = blah-queue-thankyou,blah-queue-pleasewait
By: Claudio Villalobos (devmod) 2010-06-29 18:08:53

I add the members dynamically btw. (Local extensions)
By: Tilghman Lesher (tilghman) 2010-06-29 19:51:20

And there's your problem.  Eventwhencalled is not a global setting, but a per-queue setting.  Move it down from the global section and it should work fine.  In fact, only the first 4 settings in your [general] section are actually values for the [general] section.  The others are all ignored.