similar to: FOP flash panel: how to reload config files when running

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

Hi, is it possible to remove the "no timeout" combo box in flash operator panel? How can I reduce the flash area? I set small buttons and half of the area is white and I want to resize it. TIA Giorgio Incantalupo

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, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

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

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, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

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

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.

Hi, I'm trying to use Hylafax without a modem. Is it possible to use t38modem to make Hylafax send and receive fax via Asterisk? If yes, how? I'm searching on internet but still haven't found anything useful. TIA GIorgio Incantalupo

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site),