search for: gizmos

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: Esta =?ISO-8859-1?Q?=E9?= uma parte de mensagem assinada digitalmente Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070523/441a4928/attachment.pgp

...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 voicemail off the call gets answered by a recording saying I'm not

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

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 tips at where I should be looking .... firewall port forwarding ....

...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 a...

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

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

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have 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

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have 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

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

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

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 of rows + tcn5[is.na(tcn5[n,i]) | tcn5[n,i]

When I try to execute the following example, I get an error message: /usr/local/lib/ruby/gems/1.8/gems/mocha-0.5.4/lib/mocha/object.rb:40: in `expects'': undefined method `stub'' for nil:NilClass (NoMethodError) from test8.rb:5 What could be the reason? I tried with the latest Mocha Ruby gem, and I also tried it with the Rails plugin. The example: require

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>