asterisk users - Jul 2010 - Got SIP response 603 decline, then the call hang up

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 [103 at AgentDialFromQ:2]
Hangup("Local/103 at AgentDialFromQ-eb28,2", "") in new
stack

  == Spawn extension (AgentDialFromQ, 103, 2) exited non-zero on
'Local/103 at 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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