similar to: T-Mobile and WiFi Voip

Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

Hi all, I've try to search Google about this without any chance. I want to know if it's possible to use a mobile phone application for redirect automatically incoming calls of a GSM phone connected to Wifi network to a Sip phone. I've try to use different mobile phones SIP clients without any success. No one of them can redirect calls automatically. I've got Android and

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Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

http://www.siliconvalley.com/mld/siliconvalley/business/international/as ia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and

Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before

I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The "problem" with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries.

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a

Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/a278cbbd/attachment.htm

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

I am wondering how the g729 licenses are done during calls. If I have N licenses for g729, and N are in use and an additional call comes in that requests N+1 to be in use, how does asterisk handle that call? Does it dump it? Does it negotiate another codec automagically? Basically what happens to that call, obviously it wont (shouldnt) let you use more licenses than you have available, but

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

Hi BJ, Further explanation about the 3rd party ecosystem question this morning Cory Andrews from VoIP supply was on the Voip-Users conference call last week. I asked the question - how much of VoIP Supply revenue is product versus applications - he said we don't sell any services such as ITSP hosted Asterisk so I replied that wasn't what I was thinking of and gave the example of Snap

Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of "late packets". The stats when running version *-.0.4.0 do not

DSL works by using the frequencies above 4k that were unused in POTS loops of yesterday. Load Coils, Bridge Taps, and DC taps are all devices added to lines to increase their reach and stability, unfortunately, they are DEADLY for DSL. Other problems can effect DSL service, and cause it to be 'flaky'. 1 Temperature, in Florida the large black cables are constantly beaten down by the sun,

For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in the Debian devroom on Open Source VoIP. http://www.fosdem.org/2007/schedule/speakers/daniel+pocock Several VoIP projects will be represented in various ways throughout the weekend, and there will be some of the following: - hardware giveaways from leading VoIP companies - launch of new open source VoIP product

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh