search for: gizmo

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware cards yet) X...

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...y because they don''t want to install the Ruby environment just for me... But at home, I still prefer the Ruby language... So, wxWidgets is a great library, but wxPythons add some intersting stuff. I''m specially interested by the TreeListCtrl. it seems that this control belongs to Gizmos C++ library. I''m wondering if the Gizmos library is included in wxRuby or do you plan to include it soon? Bruno.

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicema...

Got an email this morning with the subject "Welcome to Gizmo Project". I didn't sign up with those yokels. Anyone else got spammed by them?

Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda worked, but it wouldn't receive any calls. The Gizmo one worked ONLY to call their echo test - If I tried to call a real number I got some serious noise, sometimes the phone ringing,...

Hi all, I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine .... I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them .... again if the call comes INTO the server both sides work fine. Just looking for some...

...ot.org (home) -or- djm at ibs.com.au (work) ---------- Forwarded message ---------- Date: Wed, 19 Apr 2000 16:24:57 -0400 From: Don Davis <dtd at world.std.com> To: Jeff.Hodges at stanford.edu Cc: cryptography at c2.net, dtd at world.std.com, soley at sun.com Subject: Re: Entropy-gathering gizmos for Solaris? > Ok, now I'm curious about what all entropy-gathering gizmos, ... > might be available for/on Solaris -- anyone know? the solaris security toolkit has a program called cryptorand. about 3-4 years ago, i intensively reviewed and tested crypto- rand for a client. i talked...

Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder. It obviously results in hearing silence when the call is bridged (you've picked the phone). I...

Hello there, I'm having difficulties to configure Asterisk to handle multiple Gizmo accounts. Ideally, I'd like each account to go to a different context/extension. Can anyone help me out? Sample configurations are welcome!! Best, Leo

Hi All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my hosted pbx from Skype and Google Voice via the Gizmo and...

...e a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open beta: go for it. Pricing isn't yet established, but other questions may be answered today. One of our regulars, Maxim, mentioned GizmoVoice.com which explains everything you wanted to know about getting a free Google Voice number and being able to use it with SIP, thanks to Gizmo & co. I've done this with OpenSky and it has worked well so far. Join us at 12 Noon EDT, 9AM PDT, 5PM UK or: http://tr.im/vuctime for your time...

...e a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open beta: go for it. Pricing isn't yet established, but other questions may be answered today. One of our regulars, Maxim, mentioned GizmoVoice.com which explains everything you wanted to know about getting a free Google Voice number and being able to use it with SIP, thanks to Gizmo & co. I've done this with OpenSky and it has worked well so far. Join us at 12 Noon EDT, 9AM PDT, 5PM UK or: http://tr.im/vuctime for your time...

Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? "OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls." http://www.gizmo5.com/pc/opensky/ Julian ________________________________________________________________...

I just went through an exercise of writing a Perl script called from my Asterisk dialplan to look at a list of area codes and exchanges to determine which ones are local (no or little cost) under my current Verizon plan. I route calls outside of my local limits to Gizmo. It works fine but when I called Verizon to change (lower) my service it was a bewildering spider web of rates structures just in the Philadelphia metropolitan area. It made me wonder why I send any of my calls to Verizon! I was able to cut my Verizon cost down by about half. I wonder if any other...

...ent. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a requirement. ) SIP is preferred, but IAX may also work. ) Cross platform is preferred, but Windows is the primary user base. ) They want it done "yesterday." Gizmocall works fine, but requires their customers to create a gizmo account just to call my client and it does not have speed dial buttons. Any suggestions? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voi...

Hi all, Is it too soon to start porting EH code to the new C++ API? Are there sample implementations? llvm.org/demo still uses the old EH gizmos. should we wait for 3.0 release? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20110930/3bc09ea9/attachment.html>

Oops, I just erased all my data using this gizmo that I thought would replace -9 with NA. A) Can I get my tcn5 back? B) How do I do it right next time, I learned my lesson, I'll never do it again, I promise! Anders Corr > for(i in 1:dim(tcn5)[2]){ ##for the number of columns + for(n in 1:dim(tcn5)[1]){ ##for the number...

...quire ''rubygems'' require ''mocha'' some_time = Time.at(0) Time.expects(:now).returns(some_time) puts Time.now -- Felix E. Klee Jabber/Google Talk: feklee at jabber.org, SIP: 9779619 at sipgate.de ICQ: 158124695, Yahoo!: feklee, AIM: felix.klee at inka.de Gizmo: felixklee, Skype: felix.klee

...ityjumping=yes) and it started to work. If that was the problem (which it seems to be), is that the wrong default? Or am I missing something here completely? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com