similar to: OT: FOP and reverse_transfer

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

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

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 all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones. The "Flash Operator Panel" requires that we set a static value for each line or

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

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

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

Hello users! Recently I read that AgentCallbackLogin is going to be deprecated soon. Wanting to set up a few callback type queues, I set them up as suggested in queues-with-callback-members.txt. I was able to set the queues up completely this way, however, I'm trying to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login status. FOP monitors their status if I call

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

On Tue, 15 Feb 2005 03:02:45 +0100, Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I have some trouble with the FOP and would appreciate if anyone could > point me into the right direction. There is a FOP user list, although not too active. http://www.asternic.org/ > Is there a way to define a button like Zap/g1/6000 and have it light up > when

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.

Hi All, Whenever i try to start FOP using script ./op_panel_redhat.sh start given in directory /usr/local/op_panel-snapshot/init I got the following error: Starting Flash Operator Panel: execvp: No such file or directory [FAILED] Please let me know the reason for this. Thanks in Advance With Regards, newbie

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

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Hi, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. Bye, Sven -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, It seems that my flash panel only works when I specify my ip address and not the host name. I've tried quite a few things (change host file, dns resolve, proxying....) but couldn't get it to work. Anyone knows how to solve this? Thanks, Ohad -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, I know this is not a 100% Asterisk question, but is there anyone who has the Flash Operator Panel working with Asterisk 1.6?? In asternic.org there is a version that show call status but you cant make transfers or originate a call. Has anyone fixed the op_server.pl file to fully work with Asterisk 1.6??? Thank you very much, Pablo Bernasconi -------------- next part -------------- An HTML

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a