Hello, this software has worked great on a number of systems. Thanks again for programming it. I'll be moving to fop2 as soon as I can convince my company to use a supported Linux version, anyhow,

scenario:

Upgrade of asterisk from 1.4 to 1.6.2.20

Users report that status is not working as it once was

manager is connected:

CLI> manager show connected
Username IP Address Start Elapsed FileDes HttpCnt Read Write
manageruser 127.0.0.1 1313710669 172 18 0 04079 04851
1 users connected.
*CLI>

• The screen shows some extensions greyed out indicating that it has some connection.
• status is displayed on some extensions, although slowly
• incoming calls dont show the shaking phone animation, and yes it was always turned on:

; If enabled, the phone will shake/ring
enable_animation=1

UPDATE:

one server it works when
manager_host=127.0.0.1 for some reason, i guess that will do

ALSO

It is not enough to simply reload the flash panel service, reload asterisk, and reload the page.

It ONLY worked when I set the 127.0.0.1 as host and then backed completely out of the flash panel, and re-clicked the link for my desired context. About 3 hours of constant hammering at it, but now I can replicate a working system with consistency.

-----------------------------------------------------------------------------------

connection to asterisk manager is rock solid, buttons display perfectly, every asterisk context includes every other context,no flashing red and green lights, in other words it should work.

The problem is that status indication and transfers do nothing, that is to say no response or activity from the flash panel at all.

In the past I have done dozens of installs with no issues, whats wrong with this one?

asterisk-1.6.1.6
flash-plugin-10.0.32.18-release.i386

manager.conf
-------------------------------------
[general]
enabled = yes
webenabled = yes
port = 5038
bindaddr = SERVERIP
bindaddr = 127.0.0.1
allowmultiplelogin = yes
displayconnects = yes

[FLASHUSER]
secret = FLASHPASS
read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
write = system,call,agent,user,config,command,reporting,originate

op_server.cfg
-------------------------------------
[general]
; If you want to use freepbx/trixbox conf file, set this to 1
use_amportal_conf=0;

; host or ip address of asterisk
manager_host=SERVERIP
manager_port=5038
; user and secret for connecting to * manager
manager_user=FLASHUSER
manager_secret=FLASHPASS
; The optional event_mask for filtering manager events.
; Asterisk will send only the events you request
; with this parameter. Possible values are:
; on, off, system, call, log, verbose
;event_mask=call
;
; You can specify many asterisk servers to
; monitor. Just repeat the manager_host, manager_user
; and manager_secret parameters in order. The first
; one will be server number 1, and so on.
;
; manager_host=1.2.3.4
; manager_user=john
; manager_secret=doe

; Enable MD5 auth to Asterisk manager
auth_md5=1

; you can use astmanproxy, if you enable it, all of the above
; connections and settings will be overriden. You have to define
; the host and port
; astmanproxy_host = 127.0.0.1
; astmanproxy_port = 1234

; You will also have to define the servers that are monitored trough
; astmanproxy, you have to enumerate them using the astmanproxy_server.
; astmanproxy_server = 192.168.10.1
; astmanproxy_server = 192.168.10.2
; astmanproxy_server = 192.168.10.3
;
; ip address to listen for inbound connections, default all
;listen_addr=127.0.0.1

; port to listen for inbound flash connections, default 4445
;listen_port=4445

; hostname or ip address used to connect to the webserver where
; the flash movie resides (just the hostname, without directories)
; This value might be omited. In that case the flash movie will
; try to connect to the same host as the web page.

web_hostname=SERVERIP

; location of the .swf file in your disk (must reside somewhere
; inside your web root)
flash_dir=/var/www/html/panel/flash

; secret code for performing hangups and transfers
security_code=2005

; Frequency in second to poll for sip and iax status
poll_interval=12000

; Poll for voicemail status (only necesary when you access the
; voicemail directories outside asterisk itself - eg. web access)
poll_voicemail=0

; 1 Enable automatic hangup of zombies
; 0 Disable
kill_zombies=0

parkexten=700
parktimeout=30

; Debug level to stdout (bitmap)
; 1 Manager Events Received
; 2 Manager Commands Sent
; 4 Show Flash events Received
; 8 Show events sent to Flash Clients
; 16 Server 1st Debug Level
; 32 Server 2nd Debug Level
; 64 Server 3rd Debug Level
;
; Eg: to display manager events and
; commands sent set it to 3 (1+2)
;
; Maximum debug level 255
debug=0

; Default language to use (op_lang_XX.cfg file)
language=en

; Context in your diaplan where you have the conferences for barge in
; Example:
;
; meetme.conf
; [rooms]
; conf => 900
; conf => 901
; conf => 902
;
; extensions.conf
; [conferences]
; exten => 900,1,MeetMe(900)
; exten => 901,1,MeetMe(901)
; exten => 902,1,MeetMe(902)
conference_context=conferences

; Meetme room numbers to use for barge in. The room number must match
; the extension number (see above).
barge_rooms=900-902

; When doing barge ins, you can make the 3rd party to start
; the meetme muted, so the other parties wont notice they are
; now being monitored
barge_muted=1

; Formatting of the callerid field
; where 'x' is a number
clid_format=${CLIDNAME} (xxx)xxx-xxxx

; If you want not to show the callerid on the buttons, set this
; to one
clid_privacy=0

; To display the ip address of sip or iax peer inside the button
; set this to 1
show_ip=0

; It will hide queue position buttons and show only the active ones
queue_hide=0

; Will change the button label on AgentLogin
rename_label_agentlogin=0

; Will change the button label on Agentcallbacklogin
rename_label_callbacklogin=0

; Will rename the label for a wildcard button
rename_label_wildcard=0

; Will rename to the name specified in agents.conf
; If disabled the renaming will be Agent/XXXX
rename_to_agent_name=1

; Will display IDLE time for agents, as well as
; update the queue status after an agent hangs up
; the call, so you don't need to reload to get
; queue statistics
agent_status=0

; Will rename labels for queuemembers
; If you use addqueuemember in your dialplan, you
; can fake an AgengLogin event by sending it with
; the UserEvent application. Eg:
;
; exten => 25,1,AddQueueMember(sales|SIP/${CALLERIDNUM}
; exten => 25,2,UserEvent(Agentlogin|Agent: ${CALLERIDNUM});
; exten => 25,3,Answer
; exten => 25,4,Playback(added-to-sales-queue)
; exten => 25,5,Hangup
;
; exten => 26,1,RemoveQueueMember(sales|SIP/${CALLERIDNUM})
; exten => 26,2,UserEvent(RefreshQueue);
; exten => 26,3,Answer
; exten => 26,4,Playback(removed-from-sales-queue)
; exten => 26,5,Hangup
rename_queue_member=0

; Will change the led color when the agent logs in
; The color is configurable in op_style.cfg
change_led_agent=1

; If set to 1, you will transfer the linked channel instead
; of the current channel when you drag the icon on a button
reverse_transfer=0

; If enabled, it will not ask forthe security code
; when performing a click to dial
clicktodial_insecure=1

; Enable select box with absolutetimeout for the call after
; a transfer is performed within the panel
transfer_timeout= "0,No timeout|300,5 minutes|600,10 minutes|1200,20 minutes|2400,40 minutes|3000,50 minutes"

; If set to 1, when hitting the reload button on the flash
; client it will instead restart the 1st asterisk box
; (For asterisk to restart you have to start it with
; safe_asterisk, if you dont do that, asterisk will just
; shut down)
enable_restart = 0

; If you set this parameter to your voicemailmain
; extension@context, it will originate a call to
; voicemailmain when double clicking on the MWI icon
; for any button.
voicemail_extension = 3000@features

; Channel variables to be passed from origin channels to Ringing channels
; Those variables will appear in the popup base64 encoded. A new event
; will be generated to clients in the form:
; "setvar" and data VARNAME=BASE64(value)
passvars=FROM_DID

; Attendant transfers. If this parameters are uncomented, then
; barge in functionality will be replaced with attendant transfers
;
; You will need to specify special meetme extensions and another
; special hold extension. Attendant trasnfer will use the barge_rooms
; and conference_context specified above to handle the mixing via meetme
; The meetme extensions should add a priority 10 like this one:
;
; [conferences]
; exten => 901,1,Meetme(901|qMAx)
; exten => 901,2,Hangup
; exten => 901,10,Meetme(901|qMx)
; exten => 901,11,Hangup
;
; exten => 8765,1,MusicOnHold
;

;attendant_hold_extension = 8765
;attendant_hold_context = conferences

; When attendant transfer fails to originate the call to the destination
; you can specify a custom failure redirect with the parameter
; attendant_failure_redirect_to. For example, you can redirect
; the call to voicemail if the attendant fails. If this parameter is commented
; the call will be bridged back to the transferrer. In this example, if you
; try to transfer to extension 100 and it fails, the call will be transferred
; to 6100 instead (where you can have the voicemail app, or anything else,
; maybe a queue, etc).

;attendant_failure_redirect_to = 6${EXTEN}@${CONTEXT}

; It is possible to start monitoring a conversation
; by single clicking on the arrow for a button
; FOP will use a filename and format based on the
; following two paramters:

;monitor_filename = FOP-${CLIDNUM}-${LINK}-${UNIQUEID}
;monitor_format = gsm

; You can have panel contexts with their own
; button layout and configuration. The following entry
; will create a context called sip with a different
; security code. In the online documentation you will
; find how to use contexts
;
;[sip]
;security_code=djdjdi43
;web_hostname=www.virtualwebserver.com
;flash_dir=/var/www/virtualwebserver/html/panel
;barge_rooms=800-802
;conference_context=otherconferences
;transfer_timeout="0,No timeout|60,1 minute"
;voicemail_extension=1000@nine
;language=es

Wow, fast responses. Thank you for the feedback, I didnt have time to include more info

To clarify, the op_panel has been daemonized by putting the redhat startup script in init.d and doing chkconfig --add op_panel, this has worked great.

The connection to the manager interface appears true, the lights don't flash, and status notification works. If you manually dial into a meetme room, asterisk lets you in, and the red letters appear on the flash button. The manager permissions are also identical to a dozen working pbx's we have with the fop installed and fully functional.

The only problem, is that the buttons are not functional for transfers.

In this state the FOP is fine for watching what is going on, as status works, but not for actually transferring calls.

The sip extension's in op_buttons.cfg are identical to other systems, extensions.conf contexts are inclusive, etc,

/var/lock/subsys/op_server.pl -X 15

no output

The flash panel .29 worked perfectly for year or more with asterisk 1.2/1.4. After upgrading to 1.6 it just sits there uselessly.

Lights stay constant green, no errors are given, the problem is that the cli shows nothing when the buttons are manipulated.

I tried upgrading the fop to .30, which is supposed to work with asterisk 1.6, but no go.

Whats the deal?