search for: fonal

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.

...terface leaves a lot to be desired... The idea of the storing settings compact flash is nice, although sometimes they choose not to stick. All in all, we're thinking about returning it in favor of something a bit more tested (and easier on an asterisk newbie). :) Currently, potentials include Fonality's PBXtra & Switchvox PBX because they are turnkey in nature AND have ultra-friendly admin interfaces (read: highly customizable * implementation without touching the command-line - nice for a quick & powerful deployment). Anyone out there know of alternatives to either of these prod...

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 i...

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 sedwar...

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 fine, but there is some desire to get rid of the analog lines. Could one add som...

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 chang...

...s it brings you. We can only accommodate the first 1000 trixboxers on the webinar so sign up today and call in early! August 13 @ 9:00 AM PDT (16:00 UTC/GMT) <http://echo4.bluehornet.com/ct/ct.php?t=1839204&c=1686662876&m=m&type=1 &h=24C5C6D927F8E6A426872B034CFC94F4> Fonality, Inc. Fonality | 200 Corporate Pointe Suite 350 | Los Angeles, CA 90230 | www.trixbox.org <http://www.trixbox.org/> This message was intended for: dean at cognation.net You were added to the system May 3, 2007. Click here for more information <http://echo4.bluehornet.com/subscribe...

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and t...

...launch an outbound call. That would take care of integrating a WHOLE LOT of devices, as (sadly) the MS contact database would be the go-between that all of these devices synch with in one way or another already. Is TAPI the right protocol to investigate for this purpose? Would something like Fonality's HUD software bridge this gap? Has this wheel already been invented? Hoping for some thoughts! Cheers, j

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason

Is there something similar or better than HUD pro out there for asterisk PBX. HUD pro is wonderful thing, but they require complete Fonality product to be purchased first, and don't sell it as a stand alone product. If someone is not interested in Fonality product but is ready to purchase some good interface like HUD, what are the options? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrub...

Meetme cant handle more than 5 users in a call?? Hmmmm http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/ hmmm I'm all for commercializing a product, but this FUD from Fonality seems to be taking it just a little too far Regards, Dean Collins Cognation Pty Ltd dean@cognation.net <mailto:dean@cognation.net> +1-212-203-4357 Ph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...

Just found out my maintanence agreement with fonality does not cover phones over a year old. So, of course, they don't repaird phones, only replace them. Very worldly of them... Anyone know where I can get an Aastra 480i repaired? The phone works on speakerphone, but when you lift the receiver offthe hook, the phone does not engage. There...

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the li...

...configure the phone like this: [snom300] disallow=all allow=ulaw ; preferred codec allow=g729 then (1) will fail because it's trying to do this: Snom300 ---ulaw---> (transcode) ---g729----> PSTNPeer This is the best description of the problem I've found online: http://fonality.com/trixbox/forums/trixbox-forums/open-discussion/codec-selection-negotiation-and-tweaking but unfortunately it doesn't come with a solution. Is there a way to prevent my IP phone from always connecting to my Asterisk box using its preferred codec or is that simply impossible? Cheers, Fra...

...if I'm missing some other config required for this to work. Many thanks in advance, [image: Fon] <http://www.fon.com/>Tim JonesSoftware Development+34 612345678 C/ Quintanavides 15, Edificio 2, Planta 1? Parque Empresarial V?a Norte de Metrovacesa Las Tablas 28050 MadridSkype: tim.jones.fonAll information in this email is confidential <http://corp.fon.com/legal/email-disclaimer>

I've noticed that some products, like Fonality's HUD, allow live monitoring of a VoIP call (not just Zap Barge). The Asterisk {client | manager} command set only seems to allow recording to a file without the use of a meetme room. Does anyone have a good solution for this? What I'd like to implement, ideally, is that once an inco...

...l trixbox is all about FreePBX. If > >> they remove FreePBX from Trixbox, nothing is left in it. A half > >> working HUD, and another small little things don't make any major > >> difference after all. > >> > >> So are the FreePBX developers with the Fonality team or with the open > >> source community? I think now is the time for someone to come up with > >> another similar product to compete with them. Are there any such > >> people around who agree with me and can spend time and expertise to > >> develop such a...