similar to: Gizmo Dial Out No CALLED PARTY AUDIO??

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

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

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp

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

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

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?

> Anyone connected up to it yet? > > http://www.skypeforsip.com/ This service is vaporware. It's just surveyware at this point with no actual service. An alternative is OpenSky which is a launched service which does SIP to Skype and Skype to SIP so you can answer and make all your Skype calls from any SIP aware device. There's a comparison chart at: http://sipforskype.com and

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

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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

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, 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

Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters "2" we know they want bishop if they enter "336" they want demon and "332"

I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though

>> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote: >> >> > Hi there, >> > >> > is gizmo the first user of the Digium Skype solution, or do they use a >> > different approach/product - any clue? >> > >> > http://www.gizmo5.com/pc/opensky/ >> > >> > Philipp OpenSky is no related to any product from Digium.

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

Daniel wrote: For us, opensky can be OK for individual users, not for allowing Asterisk users to call Skype users. Why? Simply that when you buy the 20 USD connection to Skype and don't want your calls to be cutted after 5 mn, you have to use the Gizmo Skype aliases system which is in your account. Not really helpful if you want to connect transparently your users to Skype! They better had to

my problem is this. I have google forward the call to gizmo5. I have this line in my sip file : register => user:password at proxy01.sipphone.com I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it? At the end of my sip file i have this [Calls-From-Gizmo-Network] type=user context=demo disallow=all allow=ulaw allow=ilbc allow=gsm

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