<div>Hi Matthew,</div>
<div>BroadTel is pleased to release UPA-1, a USB to FXS adapter embedded with SIP softphone. Product specification is as follows:<br> <br>Hardware:<br>USB to RJ11 FXS adapter <br>1 USB port, for computer connection <br>1 RJ-11 FXS, for phone connection <br>
Dimension (L x W x H): 53 x 15 x 28 mm <br>Support SIP 2.0, IAX2 or Skype <br>built-in 2MB flash drive for storage of BroadTel softphone<br>Support any analog phone or cordless phone <br>Support connecting to PBX <br>Plastic case for the adapter can be designed on a per-client basis<br>
<br>Main Features<br>USB to RJ11 FXS converter <br>Support Windows AUTORUN for SELF loading of driver and softphone <br>Embedded SIP, IAX2 softphones or Skype, supporting G.729, G.723.1, G.711(uLaw,aLaw), GSM(FR,AMR), iLBC, Speex and so on <br>
SIP and IAX softphones can be customized with customers'logo <br>Auto detection, installation and configuration of Windows USB audio deveice when UPA-1 is plugged in <br>Support standard windows USB audio device with unique device ID <br>
Connect USB phone adapter UPA-1 to your phone, no more microphone and speaker <br>SLIC interface for analog phone connection <br>Support cordless phone set include DECT, 2.4GHz, 900MHz or others <br>Receive SIP, IAX2 or Skype calls by ringing and picking up handset - same as home line <br>
Dial SIP, IAX2 and calls through phone pad directly or softphone users interface <br>Connect Skype and SkypeIn calls into PBX or enterprise IVR <br>Dial SkypeOut from PBX digital extension set directly <br>Support Skype speed dial and SkypeOut through phone pad directly <br>
Support 1 REN standard loads<br> <br>OEM and ODM orders are welcome. For details, please visit BroadTel corporate web site <a href="http://www.broad-tel.com/products/phoneadapter.php">http://www.broad-tel.com/products/phoneadapter.php</a> , or conact us at <a href="mailto:broadtel@126.com">broadtel@126.com</a> if you are interested in the product.<br>
<br>Best regards,<br>BroadTel </div>
<div><br><br> </div>
<div class="gmail_quote">On Fri, Mar 21, 2008 at 4:29 PM, Matthew Rubenstein <span dir="ltr"><<a href="mailto:email@mattruby.com" target="_blank">email@mattruby.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"> I saw an ad for MagicJack on TV, a USB VoIP dongle with a $20:year<br>unlimited calling subscription, and looked into it. I am in no way<br>
affiliated with MagicJack, except that I saw the ad and was curious. It<br>was introduced at the TED conference last April, and was buggy through<br>last Summer. But it's been around a year now, has a budget for<br>mass-market advertising.<br>
<br> Anyone know any more details? How do they offer $20:year, when most<br>VoIP competitors charge at least $15-35:month? Are they using Asterisk<br>for infrastructure - any thing more than maybe just voicemail?<br>
<br> If this isn't a scam, or a bubble loss-leader that will collapse under<br>a $20:year subscription, is it instead a new force pushing down prices,<br>and pushing ahead the mass marketization of Internet voice?<br>
<br><br> I tried to send this message directly to the list WED afternoon, but<br>once again the Digium filter is silently rejecting my posts. Digium<br>really should fix that. And soon. These lists are badly broken by it.<br>
<br><br>On Thu, 2008-03-20 at 12:00 -0500, <a href="mailto:asterisk-biz-request@lists.digium.com" target="_blank">asterisk-biz-request@lists.digium.com</a><br>wrote:<br>> Date: Wed, 19 Mar 2008 18:34:15 -0500<br>> From: "Kevin P. Fleming" <<a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a>><br>
> Subject: Re: [asterisk-biz] Asterisk - SIP - H323 - IAX<br>> To: Commercial and Business-Oriented Asterisk Discussion<br>> <<a href="mailto:asterisk-biz@lists.digium.com" target="_blank">asterisk-biz@lists.digium.com</a>><br>
> Message-ID: <<a href="mailto:47E1A2F7.7010905@digium.com" target="_blank">47E1A2F7.7010905@digium.com</a>><br>> Content-Type: text/plain; charset=ISO-8859-1<br>><br>> Seysan wrote:<br>><br>> > If yes, what have you used for H323 part? I'm not concerned about<br>
> RTP<br>> > Passing through the Asterisk Box (except maybe for IAX), and it is<br>> not used<br>> > as an User Agent.<br>><br>> IAX2 does not use RTP. Asterisk is always a User Agent. SIP and H.323<br>
> channels using RTP will always start out with RTP flowing through the<br>> Asterisk box, and based on my understanding of H.323, it is not<br>> possible<br>> to redirect the RTP media to a different endpoint once the channel is<br>
> setup.<br>><br>> In summary, while Asterisk is in a lot of ways 'like a softswitch', it<br>> is not a softswitch.<br>><br>> --<br>> Kevin P. Fleming<br>--<br><br>(C) Matthew Rubenstein<br><br>
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