Another login problem
-
Dear All.
I want to try fop2 in my elastix 2.2 (rc3) installation.
I installed FOP2 2.24 via elastix addon manager, the install went fine.
My fop2.cfg:[general] ; AMI definitions manager_host=localhost manager_port=5038 manager_user=fop2 manager_secret=au2koslz5 ; Global Config poll_interval = 86400 poll_voicemail = 1 monitor_ipaddress = 0 ; Force blind transfer on asterisk 1.6 blind_transfer = 1 master_key = 1223342 spy_options="bq" whisper_options = "w" monitor_filename=g${DEST_EXTENSION}-${UNIQUEID} monitor_format=wav monitor_mix=true voicemail_path=/var/spool/asterisk/voicemail #exec autoconfig-users-freepbx.sh/usr/local/fop2/fop2_server --test
Flash Operator Panel 2 - No valid license found. Demo Mode Connection to manager OK!CLI> manager show connected
Username IP Address Start Elapsed FileDes HttpCnt Read Write fop2 127.0.0.1 1318245380 1978 22 0 08191 08191 1 users connected.netstat -lnp |grep 4445
tcp 0 0 0.0.0.0:4445 0.0.0.0:* LISTEN 11045/fop2_serverIt seems like everything working fine, but I unable to login via web. I'u using one extension, with it's voicemail passvord.
Ins asterisk database, the fop2user table contains it:1 0 200 258258 allWhen I started in debug mode, I tried to connect I found this on the console:
/usr/local/fop2/fop2_server -X 511** MAIN New client connection 10.2.3.241 ** MAIN AMI event received... HTML5 Web Socket Connection Use of uninitialized value in substitution (s///) at script/fop2_server_2.20.pl line 9496. Use of uninitialized value in substitution (s///) at script/fop2_server_2.20.pl line 9497. Use of uninitialized value in concatenation (.) or string at script/fop2_server_2.20.pl line 9506. 10.2.3.241 => HTTP/1.1 101 Web Socket Protocol Handshake Upgrade: WebSocket Connection: Upgrade WebSocket-Origin: WebSocket-Location: ws://10.2.3.172:4445/ WebSocket-Protocol: sample ** MAIN closing 10.2.3.241 ** CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0x1bec1610) ** CLEAN SOCKET flash client connection lost No flash clients connected ** MAIN New client connection 10.2.3.241 ** MAIN AMI event received... HTML5 Web Socket Connection Use of uninitialized value in substitution (s///) at script/fop2_server_2.20.pl line 9496. Use of uninitialized value in substitution (s///) at script/fop2_server_2.20.pl line 9497. Use of uninitialized value in concatenation (.) or string at script/fop2_server_2.20.pl line 9506. 10.2.3.241 => HTTP/1.1 101 Web Socket Protocol Handshake Upgrade: WebSocket Connection: Upgrade WebSocket-Origin: WebSocket-Location: ws://10.2.3.172:4445/ WebSocket-Protocol: sample ** MAIN closing 10.2.3.241 ** CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0x1b67e820) ** CLEAN SOCKET flash client connection lost No flash clients connected ...The version show this:
/usr/local/fop2/fop2_server --versionfop2_server version 2.21 betaBut in the elastix, the package version is : FOP2 v2-2.24
When I try to log in, it shows:
Connecting to server, attempt number : 1 till 15 and than "System is not available right now"The elastix (which have asterisk 1.8.7.0) and the fop2 is on the same machine. No firewall, or anything else.
What is the problem? Please anybody help me, because I want to buy it, but if it's not working than I don't want to spend money...
Thanks
Robert
-
I am having the same problem. About 2 months ago I bought the full version and paid for the professional install. It was working fine until two days ago when I changed some extension names and added a few more. I used both the Elastix GUI and the FreePBX GUI. Now when I try to log in it times out. The version says 2.21 beta. The version doesn't appear in the Elastix about list. Show manager connections doesn't show it running. Below are the configs
Kernel
Linux(i386)-2.6.18-194.3.1.el5Elastix
elastix-2.0.0-62
elastix-firstboot-2.0.0-14
elastix-email_admin-2.0.0-23
elastix-system-2.0.0-38
elastix-conferenceroom-0.0.0-10
elastix-asterisk-sounds-1.2.3-1
elastix-vtigercrm-5.1.0-8
elastix-agenda-2.0.0-24
elastix-fax-2.0.0-18
elastix-reports-2.0.0-20
elastix-developer-2.0.0-4
elastix-sugarcrm-addon-5.2.0l-5
elastix-a2billing-1.3.0-4
elastix-addons-2.0.0-19
elastix-pbx-2.0.0-40
elastix-callcenter-2.0.0-14RounCubeMail
RoundCubeMail-0.3.1-5Mail
postfix-2.3.3-2.1.el5_2
cyrus-imapd-2.3.7-7.el5_4.3IM
openfire-3.5.1-2FreePBX
freePBX-2.7.0-9Asterisk
asterisk-1.6.2.13-0
asterisk-perl-0.10-2
asterisk-addons-1.6.2.1-0FAX
hylafax-4.3.9-0rhel5
iaxmodem-1.2.0-1.1DRIVERS
dahdi-2.3.0.1-3
rhino-0.99.3-2.beta2
wanpipe-util-3.5.14-0Username IP Address Start Elapsed FileDes HttpCnt Read Write
admin 127.0.0.1 1318966167 0 27 0 08191 08191
admin 127.0.0.1 1318966014 153 26 0 08191 08191
admin 127.0.0.1 1318966004 163 12 0 08191 08191
3 users connected.[root@elastixhost ~]# tcp 0 0 0.0.0.0:4445 0.0.0.0:* LISTEN 2903/fop2_server
[root@elastixhost ~]# /usr/local/fop2/fop2_server --test/usr/local/fop2/fop2_server --test
Unknown option: test/usr/local/fop2/fop2_server
Flash Operator Panel 2 - Valid License (7)
Connection to manager OK!Asterisk manager.conf
; AMI - Asterisk Manager interface
;
; FreePBX needs this to be enabled. Note that if you enable it on a different IP, you need
; to assure that this can't be reached from un-authorized hosts with the ACL settings (permit/deny).
; Also, remember to configure non-default port or IP-addresses in amportal.conf.
;
; The AMI connection is used both by the portal and the operator's panel in FreePBX.
;
; FreePBX assumes an AMI connection to localhost:5038 by default.
;
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1[admin]
secret = elastix456
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[general]
; AMI definitions
manager_host=localhost
manager_port=5038
manager_user=admin
manager_secret=elastix456
;event_mask=agent,call,command,system,user,dialplan; Daemon definitios
;listen_port = 4445
;restrict_host = 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)
; ${MBOX} = Mailbox of the extension/user that startend the
; recording
;
; Date variables:
; %Y 4 digits year
; %y 2 digits year
; %m 2 digits month
; %d 2 digits day
; %h 2 digits hour
; %i 2 digits minute
; %s 2 digits seconds; For elastix Monitoring Tab:
; monitor_filename=g${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
;monitor_exec=/usr/local/fop2/recording_fop2.pl; 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
; 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; 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', 'hangupself',
; 'recordself', 'voicemailadmin'
;
; user=620
all:queues
; user=621
dial,transfer,pickup:deptA
; user=622
all
; user=623
meetme,pickup
; buttonfile=buttons.cfg
; ------ END SAMPLE ------; This line is NOT commented, it executes
; the autoconfig configuration for FreePBX
#exec autoconfig-users-freepbx.shI apologize for not knowing how to post things properly. I am new to posting things on a forum. Any ideas?
Randy Buller
-
-
The service is running.[/color:308gxl8j]
[root@elastixhost ~]# service fop2 status
fop2_server (pid 16851) is running..The Java Web Console in Chrome gives me this error.
Invalid UTF-8 sequence in header value
ws close event 0
close event 0
on close reseteo a cero
intendo conectar web socket en ws://192.168.1.21.:4445I ran this command that you suggested for another forum member and got the following result.[/color:308gxl8j]
[root@elastixhost ~]# /usr/local/fop2/fop2_server -X 511
** COLLECT INCLUDES fop2.cfg , tipo server
** exec: archivo temporal de server: NEWFOPVJ0tSs
** READ SERVER CONFIG
** READ SERVER CONFIG - archivo fop2.cfg
** READ SERVER CONFIG - archivo NEWFOPVJ0tSs
** READ SERVER calling COLLECT INCLUDES archivo autobuttons.cfg context GENERAL!buttons** COLLECT INCLUDES autobuttons.cfg , tipo GENERAL!buttons
** exec: archivo temporal de botones: NEWFOP6TRyZy
** READ SERVER remove temp file server /usr/local/fop2/NEWFOPVJ0tSs
** READ SERVER calling collect_includes autobuttons.cfg** COLLECT INCLUDES autobuttons.cfg , tipo GENERAL!buttons
** autobuttons.cfg already included
** READ SERVER calling collect_includes NEWFOP6TRyZy
** COLLECT INCLUDES NEWFOP6TRyZy , tipo GENERAL!buttons
** NEWFOP6TRyZy already included
** READ BUTTONS CONFIG autobuttons.cfg para contexto GENERAL
** READ BUTTONS CONFIG NEWFOP6TRyZy para contexto GENERALBorro boton temporal /usr/local/fop2/NEWFOP6TRyZy
** GENERA CONFIG START
** GENERA CONFIG - GENERAL
** GENERA CONFIG ENDCan't listen to port 4445
If I stop FOP2 and run the command, it streams information.[/color:308gxl8j]
Randy Buller
-
Hi Randy,
To start fop2_server in debug mode you first need to stop the daemonized version:
service fop2 stop
/usr/local/fop2/fop2_server -X 511However, you mention you are using Google Chrome, to make it work with the latest Chrome you must be using FOP 2.23, check that:
/usr/local/fop2/fop2_server -v
Best regards,
-
Thanks for your reply, Nicolas, It says 2.21 beta. It was working with the version of Chrome that I have now. I am unaware chrome has upgraded it self. My version of Chrome is 14.0.835.202 m. It was also working fine with Firefox.
I am not quite sure how to upgrade FOP2 and, if I do, will it overwrite my data? I will try and figure out how to do the update.
You don't think that I have a problem with port 4445?
Thank You
Randy Buller
-
If you do not want to upgrade, then use Firefox instead of Chrome.
There was another upgrade in Chrome tonight or so, and it also changes the websocket handshake so I will have to release a new version soon. Websockets is a fast moving target.
So try with Firefox and not Chrome to test connections. Remember that you must have adobe flash player installed, working and not filtered when using Firefox/Internet Explorer.
Best regards,
-
If you do not want to upgrade, then use Firefox instead of Chrome.
There was another upgrade in Chrome tonight or so, and it also changes the websocket handshake so I will have to release a new version soon. Websockets is a fast moving target.
So try with Firefox and not Chrome to test connections. Remember that you must have adobe flash player installed, working and not filtered when using Firefox/Internet Explorer.
Best regards,
-
Nicolas,
It does work. I found that it works only if Chrome is not being used for anything. I was told at the install that Chrome is the browser of choice for FOP2 and that is why I am using it. Apparently Chrome updated it self and now FOP2 will no longer work with it. I like your product but it seems that this is a weak point in the design in that browser upgrades that happen continuously will render it inoperable. I don't really care about Chrome one way or another, but the thought arises that the same thing could happen with Firefox and Internet Explorer. This will have to make for unhappy customers.
I have been scouring the forum for answers to my many questions This is a good resource and I have learned much. One of the things was about upgrading to the latest FOP2 in Elastix. I bring this up here because it seems that goes along with making Chrome work. As there is no rpm it appears to be a manual upgrade. I read your instructions. They seem simple enough but you have a warning about breaking some configs. I certainly don't want to mess my installation up as this is a demo that I use to test and demonstrate to potential clients, and I have invested paid support to make sure the install was done correctly.
I tried to download the upgrade file and it wouldn't download. Does the upgrade need to be purchased? Will my current license be applied? Your instructions require me to select the correct distro but the exact file name is not given on the download page. That is why I was trying to do the download to my computer.
I am not trying to give you a hard time because I realize that you are a one or two man operation and that it is hard to keep up with all the changes that are constantly happening with the dependent applications. I am one of the little guys too so I can relate. You are very responsive and I appreciate all your help.
All the best
Randy -
Hi Randy,
Thanks for the feedback. Believe me that I am not happy with Chrome updates either, but I cannot control Chrome releases or changes. You can still use an older chrome build, but not sure if they are available for download or not.
FOP2 is using state of the art technology. That is the price to pay if you use state of the art tech, Anyways, I could add a configuration switch to force the use of adobe flash, perhaps I will add that on the next release. That will permit you to use chrome but not taking advantage of the much faster html5 websockets implementation.
If a browser adds ore removes or changes support for some feature it is not my responsibility. If a user updates a browser, it is not my responsibility, If the browser updates itself, it is still not my responsibility. You have choices, you can still use other browsers.
What I will have to do is to streamline the work with Elastix so I can push new rpms at the same time I release new versions. That involves also a lot of additional work, packaging software is not something easy and it is very time consuming. I will try to do that in the next week, before Elastixworld.
Best regards,
Hello! It looks like you're interested in this conversation, but you don't have an account yet.
Getting fed up of having to scroll through the same posts each visit? When you register for an account, you'll always come back to exactly where you were before, and choose to be notified of new replies (either via email, or push notification). You'll also be able to save bookmarks and upvote posts to show your appreciation to other community members.
With your input, this post could be even better 💗
Register Login