Displaying 20 results from an estimated 9000 matches similar to: "Flash pannel: time display"
2005 Mar 15
2
Flashpannel: How to get more than 28 buttons?
I have setup flash pannel, ... looks nice, but so far I could not
configure it to get more than 4x7 buttons.
I tried to make the buttons smaller, but than just the entire picture is
smaller.
The description says you can have a hundred buttons, ....
Can I have multiple flash pannels? E.g. for each department?
bye
Ronald
2006 Jun 24
5
ASTCC: How to reset periodically all "card in use" flag back?
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
2006 Feb 16
2
Install instructions for FOP Flash Operator Panel do not make sense...
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
2006 May 09
3
Announcement: FOP 0.26 released
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
2005 Feb 14
1
Flash Operator Panel - lots of problems
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
2008 May 15
1
Problem while running Flash Operator Panel
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
2008 Jan 17
1
AddQueueMember and Flash Operator Panel
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
2006 Mar 29
7
Reporting?
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?
2005 Jun 23
4
Monitoring Sirrix quad BRI channels
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
2006 Mar 29
1
OT: FOP and reverse_transfer
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
2006 Nov 23
1
FOP is not displaying all my SIP extensions neither all E1 channels , why?
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
2006 Mar 20
2
pickup a call in queue
Hello,
We are faced with a problem concerning queues.
When we have several calls in different queues, is there some sort of
way to open a channel between a (sip-)phone and a SPECIFIC call in a
queue using the Asterisk manager api?
We would like to do this even when we are not a member of that specific
queue.
Thanks in advance for any suggestions!
cheers
2008 Apr 03
12
Web page to show online extensions?
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
2004 Nov 24
1
Busy Lamp Field
Some days ago there was a subject regarding BLF (SIP Phone-receptionist
Setup).
We are the developers of a Price Verify Terminal for a French company.
We have developed the hardware (small board based on a PPC 823e),
working with Linux embedded (based on Wolfgang Denk's work).
I think that it can be a good BLF.
Probably it is possible to integrate the Nicolas's FOP or a new application.
2008 Feb 27
3
Attended transfers through a GUI
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and if so, would you mind
sharing how you did it please?
Alternatively, are there any other GUIs (free or commercial) that reliably
support attended transfers?
I'm trying to
2005 Feb 08
3
live monitoring (SIP only)
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
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2005 Aug 17
4
XML Revisited
Hello Guys.
I recently contacted polycoms tech support asking if their phones supported
XML pushed information to which they replied that only model 600 had a
microbrwoser capable of reading dhtml files and such.
My question to the community is: is somebody doing any XML info push to any
brand of phones except Cisco? How are you doing it?
One of the wonders of VoIP should be the means to send
2005 Feb 01
4
astGUIclient users should not upgrade to Asterisk 1.0.5
Hello,
Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find ways
around the new callerID rules that the asterisk developers have put in place
and hope to have
2006 Feb 23
5
mpg123 alternative?
Been using mpg123 for moh for the last two years or so. However, when
I have * config errors, often times get a endless stream of console
messages and need to kill the two mpg123 processes.
Is there an alternative to mpg123 that eliminates that issue?
I see references in musiconhold.conf relative to madplay, native file
format, asterisk-addons, etc. Not sure why the asterisk-addon approach
2005 Feb 10
1
really easy FOP asterisk@home question
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.