Hello

I'm having trouble upgrading my FOP2 server from 2.22 to 2.23 when I try to execute the upgrade command I get ask for my license which is from january 19 2011

but I get this after I use the command /usr/local/fop2/fop2_server --upgrade

---------------------------------------------------------------------------------------------------
The following is the list of license upgrades available. You can buy license upgrades
codes from http://www.fop2.com/buy.php.

1. Voicemail Explorer
2. IM Chat

Enter your license upgrade code: XXXXXXXXX

There was a problem with the code, please be sure you typed it correctly.
---------------------------------------------------------------------------------------------------

do you have any ideas

thank you very much

I bougth a licence for fop2 some time ago I managed to intalled and configure it and most of the time works great. But sometimes it fail to listen and to listen and whisper for some agents, as a temporary solution I usually restart the fop2 service and it works well for a while. Please help me find where the problem migth be or what I am doing wrong. This is a big concern because for the same configuration I have diferent results sometimes I get to liston to a call to all the agents and somtime it just won-t trigger the call.

here is my configuration the fop2 is installed on an asteris 1.4.41

fop2.cfg
------------------------------------------------------------------

[general]
; AMI definitions
manager_host=192.168.0.245
manager_port=5038
manager_user=asterisk
manager_secret=**********
;event_mask=agent,call,command,system,user,dialplan

; Daemon definitios
;listen_port = 4445
;restrict_host = http://www.asternic.org
;web_dir = /var/www/html/operator/fop2

; Global Config
poll_interval = 86400
poll_voicemail = 1
monitor_ipaddress = 0

; Force blind transfer on asterisk 1.6
blind_transfer = 1

; Force supervised transfer on asterisk 1.4
; requires the atxfer manager backport patch
; supervised_transfer = 1

; Force delimiter for asterisk applications
; force_parameter_delimiter = ","

; When adding or removing members to a queue, fop2 will default to
; AddQueueMember/RemoveQueueMember commands. If you set use_agentlogin
; to 1, together with the QueueChannel in a button definition set to
; an Agent number it will use AgentCallbackLogin and Agentlogoff instead.
;
use_agentlogin = 1

; Master Password that overrides any individual one
;master_key = 5678

; Options to send to chan_spy when doing a Listen action
; This global setting is overriden by the individual button
spy_options="bq"

; Options to send to chan_spy when doing a Whisper action
; In Asterisk 1.6.1 or higher you can use B to enable barge (speak
; to both channels on a call).
whisper_options = "w"

; Filename to use when start monitoring, you can use ${UNIQUEID},
; ${ORIG_EXTENSION}, ${DEST_EXTENSION}
; and date formats %Y %m %d to construct the filename.
;
; Settings for modifying the recording filename
; Available variables are:
; ${UNIQUEID} = Unique Id of the call
; ${TIMESTAMP} = Unix Timestamp when the recording was initiated
; ${DEST_EXTENSION} = Target extenstion being monitored
; ${ORIG_EXTENSION} = Extension/User that started the recording (not
; the other leg)
; ${MBOX} = Mailbox of the extension/user that startend the
; recording
;
; Date variables:
; %Y 4 digits year
; %y 2 digits year
; %m 2 digits month
; %d 2 digits day
; %h 2 digits hour
; %i 2 digits minute
; %s 2 digits seconds

monitor_filename=g${DEST_EXTENSION}-${UNIQUEID}
monitor_format=wav
monitor_mix=true

; You can specify a script to be executed when the recording
; is finished. It will receive 3 parameters, the complete
; path and filename of the IN leg, the OUT leg and the final
; recording NAME. You should run soxmix in your script to join
; Path to your voicemail directory
; For voicemail to work the fop2 server must run on the same server
; as asterisk, or your voicemail directory must be network mounted
voicemail_path=/var/spool/asterisk/voicemail

; By default IM chats are not logged/saved. If you uncomment
; the following parameter, all chats will be stored on the chatlog
; table inside the fop2settings.db sqlite database.
;
;save_chat_log=1

; --- SAMPLE GROUPS ---
;group=queues:QUEUE/100,QUEUE/101
;group=deptA:SIP/100,SIP/101,SIP/102
; --- END SAMPLE ---

; --- SAMPLE USER LIST ---
; format: user= EXTENSION : SECRET : PERMISSIONS : GROUPS
; You can enumerate several permissions and groups separated by comma
; available permissions: 'all', 'dial', 'hangup', 'meetme', 'pickup',
; 'record', 'spy', 'transfer', 'whisper',
; 'queuemanager', 'queueagent', 'phonebook',
; 'chat', 'preferences'
user=100:1234:dial,transfer,spy,whisper,hangup,pickup,record:team
user=107:8843:dial,transfer,spy,whisper,hangup,pickup,record:team

;user=620:1234:all:queues
;user=621:1234:dial,transfer,pickup:deptA
;user=622:1234:all
;user=623:1234:meetme,pickup
buttonfile=buttons.cfg
; ------ END SAMPLE ------

==================================================================
buttons.cfg
------------------------------------------------------------------

[QUEUE/electrolux]
type=queue
label=Electrolux
context=from-sip
extension=99

[DAHDI/1]
type=trunk
label=Lineas
channel=DAHDI/1
channel=DAHDI/2
channel=DAHDI/3
channel=DAHDI/4
channel=DAHDI/5
channel=DAHDI/6
channel=DAHDI/7
channel=DAHDI/8
channel=DAHDI/9
channel=DAHDI/10
channel=DAHDI/11
channel=DAHDI/12
channel=DAHDI/12
channel=DAHDI/14
channel=DAHDI/15
channel=DAHDI/16
channel=DAHDI/17
channel=DAHDI/18
channel=DAHDI/19
channel=DAHDI/20
channel=DAHDI/21
channel=DAHDI/22
channel=DAHDI/23
channel=DAHDI/24
channel=DAHDI/25
channel=DAHDI/26
channel=DAHDI/27
channel=DAHDI/28
channel=DAHDI/29
channel=DAHDI/30
channel=DAHDI/31
channel=DAHDI/32
channel=DAHDI/33
channel=DAHDI/34
channel=DAHDI/35
channel=DAHDI/36
channel=DAHDI/37
channel=DAHDI/37
channel=DAHDI/38
channel=DAHDI/39
channel=DAHDI/40
channel=DAHDI/41
channel=DAHDI/42
channel=DAHDI/43
channel=DAHDI/44
channel=DAHDI/45
channel=DAHDI/46
channel=DAHDI/47
channel=DAHDI/48
channel=DAHDI/49
channel=DAHDI/50
channel=DAHDI/51
channel=DAHDI/52
channel=DAHDI/53
channel=DAHDI/54
channel=DAHDI/55
channel=DAHDI/56
channel=DAHDI/57
channel=DAHDI/58
channel=DAHDI/59
channel=DAHDI/60
channel=DAHDI/61
channel=DAHDI/62
channel=DAHDI/63
channel=DAHDI/64

[sip/super]
type=extension
context=from-sip
extension=100
label=Supervisor

[sip/call_1]
type=extension
context=from-sip
extension=101
label=Agente 1

[sip/call_2]
type=extension
context=from-sip
extension=102
label=Agente 2

[sip/call_3]
type=extension
context=from-sip
extension=103
label=Agente 3

[sip/call_4]
type=extension
context=from-sip
extension=104
label=Agente 4

[sip/call_5]
type=extension
context=from-sip
extension=105
label=Agente 5

[sip/call_6]
type=extension
context=from-sip
extension=106
label=Agente 6

[sip/call_7]
type=extension
context=from-sip
extension=107
label=Agente 7

[sip/call_8]
type=extension
context=from-sip
extension=108
label=Agente 8

[sip/call_9]
type=extension
context=from-sip
extension=109
label=Agente 9

[sip/call_10]
type=extension
context=from-sip
extension=110
label=Agente 10

[sip/call_11]
type=extension
context=from-sip
extension=111
label=Agente 11

[sip/call_12]
type=extension
context=from-sip
extension=112
label=Agente 12

[sip/call_13]
type=extension
context=from-sip
extension=113
label=Agente 13

[sip/call_14]
type=extension
context=from-sip
extension=114
label=Agente 14

[sip/call_15]
type=extension
context=from-sip
extension=115
label=Agente 15

Thank you very much

Jaime Infante

Hello

I've been trying to configure my licenced copy of fop2 and I almost succeeded but when I tried to listen a conversation or whisper It only works after a server restart and it does all it should do but after a while operating it stops generating the call for the extension configured. I've checked the forum for similar post with no success. My asterisk version is 1.2 please help me.

thank you very much

Jaime Infante