similar to: FOP (or equivalent) and timers

Hi all, I try to install FOP. It's very nice. In documentation I red that from my dial plan I can launch a popup window with UserEvent() application. I try to follow FOP documentation but I can't popup anything. My structure is: - server 1: Asterisk system - server 2: FOP system - client On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP

Hello, I have downloaded and installed Flash Operator Panel. version 0.21. It works pretty well and I have some questions about it. 1. The text label of the buttons are partially hidden by their icons. Is there a way to adjust right margin for the buttons? 2. I would like to have the fop brought in the front of screen whenever and extension rings. Sort of crm feature but with fop and not

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

Hi, Is it possible to turn off the request for a security code when transferring in FOP (Flash Operator Panel)? If not can the security code be set to use the SIP or voicemail passwords? I know there is a forum for FOP but no one seems to be answering there... so I thought I would see if anyone here might have experience with FOP. Thanks

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

That's what I thought it used to be but it isn't working now Here is what I have in my op_buttons.conf [801] ; Meetme must be defined by its room number Position=15 Label="MeetMe 801" Extension=801 Context=rooms Icon=5 [802] Position=16 Label="Meetme 802" Extension=802 Context=rooms Icon=5 This is in the meetme.conf [rooms] #include

Dear all, I'm happy to announce the Flash Operator Panel 0.23 release. FOP is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in AMP, Asterisk@home, etc. You can grab the latest version from http://www.asternic.org The incomplete list of new features

Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo

When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the called party gets transferred rather than the calling party. This is controlled by the reverse_transfer parameter in op_server.cfg but the behavior is exactly the same whether the parameter is set to 0 or 1. This is after the call is picked up by

Hi, I just want to deny FOP originated calls in TRIXBOX. All remaining operations (hanging up, transferring, etc) should go. Where is that option in TRIXBOX (already googled, nothing, checked config files but cant find that option)? Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica

Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

On Mon, Mar 5, 2018 at 9:19 AM, Amar Tumballi <atumball at redhat.com> wrote: > Pranith, > > > >> We found that compound fops is not giving better performance in >> replicate and I am thinking of removing that code. Sent the patch at >> https://review.gluster.org/19655 >> >> > If I understand it right, as of now AFR is the only component

Guess you went through user lists and tried something like this already http://lists.gluster.org/pipermail/gluster-users/2018-April/033811.html I have a same exact setup and below is as far as it went after months of trail and error. We all have somewhat same setup and same issue with this - you can find same post as yours on the daily basis. On Wed, Apr 11, 2018 at 3:03 PM, Anastasia Belyaeva

Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but they do not tell you were to put this file. There is something wrong with the

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

The hype and documentation for the last couple of releases of the Flash Operator Panel claim that the Panel can be configured to either change the LED for a phone, or the name of a phone to indicate when that phone is logged into a queue. I've tried on two different versions (0.18 and 0.19) on two different systems to get this feature to work, and have been completely unsuccessful. Any hints

I deleted the config examples in the op_buttons.conf folder for how to set up the meetme representation All of my other representations work fine except for the meetme meeting rooms (I know they worked in the past) and the meeting rooms themselves actually work fine just not the representation. Can anyone take a quick look at theirs and tell me what I've done wrong.