Hi to all, I have a strange behavior in my asterisk server.
I have a queue for 5 agents, the calls enter the queue an go to the agents
normally, but if I need to transfer or dial directly to an agent extension
that is already in a call, the pbx hung up the actual call (not the
transferred call).
This is what I see in the log.
Called 103
-- Agent/103 is ringing
-- SIP/103-00000e89 is ringing
-- Got SIP response 603 "Decline" back from 192.168.215.104 // (104
is the IP of SIP/103)
== Spawn extension (cola-radio2, s, 4) exited non-zero on 'DAHDI/6-1'
-- SIP/103-00000e89 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@agentdialfromq:2]
Hangup("Local/1...@agentdialfromq-eb28,2", "") in new stack
== Spawn extension (AgentDialFromQ, 103, 2) exited non-zero on
'Local/1...@agentdialfromq-eb28,2'
-- Hungup 'DAHDI/6-1'
As you can see when dialing to SIP/103 I got 603 and the actual call hung
up.
This is my queues.conf and agents.conf
[general]
;monitor-type=MixMonitor
;##################################
persistentmembers=yes
autofill=yes
joinempty = strict
leavewhenempty = strict
;##################################
[cola-radio]
musicclass = default
joinempty = strict
leavewhenempty = strict
;###################################
reportholdtime = no
ringinuse = no
strategy = rrmemory
timeout=15
retry=0
wrapuptime=1
maxlen=6
servicelevel = 60
memberdelay = 0
timeoutrestart = no
;###################################
announce=beep
announce-frequency = 30
announce-holdtime = yes
periodic-announce-frequency=10000
;periodic-announce=cu_periodic_announce
;periodic-announce=/var/lib/asterisk/cus_sounds/cu_periodic_announce
context = ivr-cola-radio
;monitor-format = gsm
;monitor-type = MixMonitor
;monitor-join = yes
;Queue Members
member => Agent/101
member => Agent/103
member => Agent/104
member => Agent/105
member => Agent/106
member => Agent/109
;member => Agent/110
member => Agent/111
member => Agent/112
member => Agent/113
member => Agent/114
member => Agent/115
member => Agent/116
member => Agent/117
member => Agent/118
member => Agent/119
member => Agent/120
AGENTS.CONF
[agents]
; Enable recording calls addressed to agents. It's turned off by default.
recordagentcalls=yes
;
; The format to be used to record the calls: wav, gsm, wav49.
; By default its "wav".
recordformat=wav
;
; The text to be added to the name of the recording. Allows forming a url
link.
;urlprefix=http://localhost/calls/
;
; The optional directory to save the conversations in. The default is
; /var/spool/asterisk/monitor
savecallsin=/var/spool/asterisk/monitor/Qcabina
ackcall=no
persistentagents=yes
;musiconhold=default
;###############################
autologoffunavail=yes
wrapuptime=1000
;###############################
agent => 101,,Operador 1
agent => 103,,Operador 3
agent => 104,,Operador 4
agent => 105,,Operador 5
agent => 106,,Operador 6
;agent => 107,,Operador 7
;agent => 108,,Operador 8
agent => 109,,Operador 9
;agent => 110,,Operador 10
agent => 111,,Operador 11
agent => 112,,Operador 12
agent => 113,,Operador 13
agent => 114,,Operador 14
agent => 115,,Operador 15
agent => 116,,Operador 16
agent => 117,,Operador 17
agent => 118,,Operador 18
agent => 119,,Operador 19
agent => 120,,Operador 20
AND THE INTERESTING PART IN DIALPLAN
-To log into the queue
exten =>
*402,1,AgentCallBackLogin(${CALLERID(num)}||${CALLERID(num)}...@agentdialfromq)
-the AgentDialFronQ context
[AgentDialFromQ]
exten =>_1XX,1,Dial(SIP/${EXTEN},,tTr)
exten =>_1XX,n,Hangup
What can be the problem?
Thanks for any help.
Ricardo
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