search for: skypein

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 wireless USB phone ? Does it support SIP and could it be used to connect to any SIP server ? Does anyone have experience with these ? Thanks. -- Kim Lux, Diesel Research I...

...can Golicher -- Dr Duncan Golicher Ecologia y Sistematica Terrestre Conservaci?n de la Biodiversidad El Colegio de la Frontera Sur San Cristobal de Las Casas, Chiapas, Mexico Email: dgoliche at sclc.ecosur.mx Tel: 967 674 9000 ext 1310 Fax: 967 678 2322 Celular: 044 9671041021 United Kingdom Skypein; 020 7870 6251 Skype name: duncangolicher Download Skype from http://www.skype.com

I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and

(Rails 1.1, Ruby 1.8.4) In the layout for one of my controllers, I have a tag to display a flash message (copied from AWDWR), like this: <div id="data"> <% if @flash[:notice] -%> <div id="notice"><%= @flash[:notice] %></div> <% end -%> <%= @content_for_layout %> </div> If the

Hello, I've seen this question asked in the archives but no clear reply or solution provided. So, just to be sure it is not possible in R: Can I replace backslashes with slashes in a string ? I am writing a GUI for R with the Rpad library. I have a "browse" button for data loading and Windows return a path string with backslashes. I need to convert them into slashes to use the

...d a codec > >> translation path from 0x100 (g729) to 0x8 (alaw) > >> > >> on the console. > >> > >> Fired up a sip client, made the same call, and all was ok. > >> > >> Any clues ? > > > > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for > > nearly all calls, so handling calls via those paths requires a G.729 > > transcoder on the system if the target of the call will not also be > > using G.729. This is why the Skype For Asterisk license includes > > licenses for Digium's G.72...

IMHO! I just see a skype channel as something good for asterisk. Skype has broad coverage. I can't imagine that skype wouldn't be interested in selling corporate accounts "skype trunk lines". Imagine having unlimited or X amount of continious calls coming in on SkypeIN and out on SkypeOUT from Asterisk. Internal Phones would all talk IAX or SIP to asterisk and use all PBX features of Asterisk. Asterisk could do lowest cost routing for internal to external calls through dundi or skype lookups (or enum later on) or through your PSTN hardware. Skype would really be...

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

Good news for service providers in my opinion, Ebay will likely start alienating Skype users like they did with Paypal users. "SkypeSucks.com" domain already taken, shucks.... http://www.kesq.com/Global/story.asp?S=3837895 -- Cory J Andrews Partner / Purchasing +++++++++++++++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++++++++++++++ tf voice