FOP monitoring multiple servers

[snowspeeder]

Can anyone help me setup FOP to monitor multiple servers? I've tried following the documentation on FOP's website, however I keep screwing something up and FOP never connects "Even the local side breaks until I enable use_amportal_conf=1." It just flashes Red and Green...

Say I wanted to add the PBX 192.168.1.100 "local" and 192.168.2.100 "Remote".

How should I modify this config file?


op_server.cfg
; NOTE
;
; In this config file, the following entries will be IGNORED, in
; favour of the values set in /etc/amportal.conf (but do not
; comment them !):
; manager_user, manager_secret, web_hostname, security_code, flash_dir


[general]
use_amportal_conf=1
; host or ip address of asterisk
manager_host=127.0.0.1

; user and secret for connecting to * manager
manager_user=UNUSED
manager_secret=UNUSED

; 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

; 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=UNUSED

; location of the .swf file in your disk (must reside somewhere
; inside your web root)
flash_dir=UNUSED

; secret code for performing hangups and transfers
security_code=UNUSED

; Frequency in second to poll for sip and iax status
poll_interval=120

; Poll for voicemail status (only necesary when you access the
; voicemail directories outside asterisk itself - eg. web access)
poll_voicemail=1

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

; 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=(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

; 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 = *97@app-vmmain

; 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