NOTHING SHOWS? or just Dahdi? Do you have sip extensions? Do you see any activity for them? IF nothing shows, you most probably have problems with your asterisk manager settings.
/usr/local/fop2/fop2_server --test
That will test the manager connection. Also be sure you have all permissions in /etc/asterisk/manager.conf
I see buttons of my extensions but no activity at all. I do not have any sip extensions. I do have some fake extensions that I am using for forwarding to cell phones using follow me but they are simple.
Here is my manager.conf (user/pass changed).
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[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+
[admin]
secret = ******
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
#include manager_additional.conf
#include manager_custom.conf
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There is nothing in manager_additional.conf and manager_custom.conf.
For u/p comparison, here is my fop2.cfg
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[general]
; AMI definitions
manager_host=0.0.0.0
manager_port=5038
manager_user=admin
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 = 0
; 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
; spyoptions directive if set (in the button config).
; Asterisk 1.6.1 or higher has the option "d" that lets you
; switch spying modes using the keypad:
;4 = spy mode
;5 = whisper mode
;6 = barge mode
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"
; When you spy to an ongoing call, your spy session will end as
; soon as the conversation you are listening to finishes. If you
; rather keep the chan spy session open after the call end, uncomment
; the following line.
;persistent_spy=1
; 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
; ${CLIDNUM} = Callerid or Dialed number for the active call
; ${CLIDNAME} = Callerid name for the active call
; ${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 started the
; recording
; ${FOP2CONTEXT} = FOP2 Panel Context
;
; Date variables:
; %Y 4 digits year
; %y 2 digits year
; %m 2 digits month
; %d 2 digits day
; %h 2 digits hour
; %i 2 digits minute
; %s 2 digits seconds
; For elastix Monitoring Tab:
; monitor_filename=g${DEST_EXTENSION}-${UNIQUEID}
; For fop2 recording interface
monitor_filename=/var/spool/asterisk/monitor/${ORIG_EXTENSION}_${DEST_EXTENSION}_%h%i%s_${UNIQUEID}
monitor_format=wav
monitor_mix=true
; To enable the recording interface you must uncomment the following
; line, but also you might need to modify the script a little bit
; depending on the sox version you have installed.
;
;monitor_exec=/usr/local/fop2/recording_fop2.pl
; You could specify your own 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
; the recordings into one file.
;
; monitor_exec=/var/lib/asterisk/bin/postrecording-script.sh
; FOP2 can fire notifications/popups when an extension or queue
; member receives a call. The default behaviour is to show a
; notification on state RINGING (notify_on_ringing=1).
;
; To customize notifications, you must uncomment the custom_popup
; function in checkdir.php you can replace that notification with
; a custom popup function to integrate with other web applications.
;
; For call centers you might need to perform a popup not on the
; RINGING state but when the call is CONNECTED to an agent. If you
; set in the queue configuration in queues.conf the option
; eventwhencalled=yes and then set here notify_on_connect=1,
; fop2 will send notifications on queue connected calls
; during AGENTCONNECT events. This will only work for inbound calls
; from a queue.
;
; notify_on_ringing = 1
; notify_on_connect = 1
; Call pickup uses the pickupmark variable by default. In multi tenant
; systems this might lead to problems as you might end un picking up
; some other tenant call. In that case you might want to try to
; pickup the call by its context uncomenting the following line:
;
; no_pickupmark=1
; If your asterisk version supports the pickupchan application it is
; much better to use this than the regular pickup application as it will
; be directed towards the channel and not the extension, makeing it
; more precise.
;
; use_pickupchan=1
; 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
; Khomp GSM interface to send SMS messages
; If there is a card plugged, fop2 will auto discover it
; and use the first one available. If you want to change it
; to a fixed one, uncomemnt the folowing line and change the name
; to your liking
;
; khomp_gsm=Khomp/b0
; --- 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', 'hangupself',
; 'recordself', 'voicemailadmin'
;
; 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 ------
; This line is NOT commented, it executes
; the autoconfig configuration for FreePBX
#exec autoconfig-users-freepbx.sh
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