similar to: FOP is not displaying all my SIP extensions neither all E1 channels , why?

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

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, 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

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.

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?

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

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

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.

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

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

Hi, In short - Asterisk is not able to recognize that the 'other' person to whom call was made has hung up - hence the channel stays busy. In long: I've been able to configure asterisk 1.1.12 with freepbx 2.1.3 and Digium TDM400 card (4 FXO ports). I've terminated 2 PSTN lines on these ports and making outbound calls successfully using the outbound rules. However, if

I have three different time displays: Flash panel caller 615 48:00 called 620 58:18 Snom phone shows for the same call 47:55 Why is there a difference at all? bye Ronald

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

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

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

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

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

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

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: