similar to: GUI interface to manage business edition

Hmm beginning of the end of free trixbox by the sounds of it. It was good while it lasted but time to download the latest iso while it's still available by the sounds of it. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: trixbox

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

Browsing through the new website... * Q - Does Asterisk Business Edition contain any additional features, fixes, or enhancements not found in the open source versions of Asterisk? * A - Digium remains committed to the open source model, and has based Asterisk Business Edition entirely on the open source tree. However, no single release of the Open Source version corresponds

Hi all.. Im using ABE C3.2.1 version and here Im having the following issue with respect to T38 fax calls. Somehow asterisk when it receives Session modification request with the T38 Codec, is not forwarding the request to the other end. Also asterisk is responding with 200 Ok to this RE INVITE with an audio line rather an media line with image. The session id here strangely matches to that of

Hello, any body know the real difference between the BE and the free one? Regards, al Andr? Lepage Directeur g?n?ral STS VoIP, Network security, Anti-virus. URL: http://www.sts.ca T?l?phone: Qu?bec 418-521-2347 Poste 227 Montr?al 514-875-1105 Poste 227 Sans frais 877-330-3305 Poste 227

A set of scripts were recently discovered in the trixbox line of PBX products, which connect to a remote host every 24 hours, to retrieve an arbitrary list of commands to be executed locally. These scripts were added under the guise of submitting 'anonymous usage statistics', however, with the help of DNS pollution, or malice on the part of the sponsoring company (Fonality), all

Looks like it's time to fork...... > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Lenz > Sent: Monday, August 13, 2007 7:28 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] FW: The trixbox Revolution Continues! Sign >

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

Hi, I am very new to this community. I want to make a custom iso which contains CentOS5.2 & scopserv(asterisk based PBX). Please provide me the proper steps for creating an iso. I have created an iso but during installation, only CentOS5.2 rpms get installed & scopserv rpms doesn't install. Is genhdlist command is required? But this command is not working with latest

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

Looks like some nice new features in fonality with their find me/follow me functionality http://www.voip-news.com/feature/fonality-find-follow-feature-030408/ love the 'boss overide' call routing - way overdue. I haven't seen the UI on this yet so cant comment on how well they've implemented it but nice to see someone implementing intelligent user controlled call routing into

An Elastix client saw a Fonality HUD demo and fell in love. I'm not a fan of Fonality as a company, but what do you think of HUD? 1) Does it bring real value? 2) Do I have alternatives? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline

Dear sir, Thanks for your reply. I apologize for taking a few days to reply. > Crashed how? Please see the below. > Have you turned on ASan? Not yet, but thanks for the suggestion. I guess I will try to rebuild my modified Clang+LLVM with addr. san. and see what happens. > I recommend attaching a minimal reproduction... Well, since I`m hacking on LLVM itself, this is not

Dear all, In the process of trying to add optimization for better layout of global variables, I have run up against a roadblock: I don`t seem to be able to sort the contents of a "SymbolTableListTraits<GlobalVariable>" -- or even swap two elements in that list -- without causing LLVM to crash. I have tried writing a comparator class and then using "llvm::iplist<

Hi All, Can anyone clerify that HUD3 is a fonality product, tied into the various trixbox systems? Is there a HUD3 client/server standalone project that can be installed and used with other Asterisk projects? Any comments on using the hudlite client/server package that came out a few years ago? Thanks. JR -- JR Richardson Engineering for the Masses

Hi, Kerry Garrison from Fonality will be with us live to address the Trixbox so-called "phone home" script issue. I'm going to try to have something about the year "2007 in review" for any and all VOIP and Asterisk-related events, so if anyone wants to report on what they've been doing in 2007, you're welcome to chime in. We're also talking about doing a

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.

Hello, I am trying to mount a lustre filesystem thru an ib2ip gateway. The MDS''s have infiniband connections. The client nodes are tcp/ip connections. I am able to route between the client nodes and the MDS''s. I have the following in /etc/fstab: abe-mds1 at o2ib0,abe-mds2 at o2ib0:/home/client /abehome lustre _netdev,flock 0 0 I get the following when trying

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works