similar to: Optimizing this script for calling Skype users from Asterisk

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

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

>> 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.

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

> 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

From: "Guillermo Salas M." <gsalas at manta.telconet.net> > http://www.gizmo5.com/opensky Free calls are available up to 5 > minutes. If you need longer calls there's a commercial service you can > purchase. > Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to

2009/3/27 Marco Sambo <derwidtel at gmail.com> > I have to try Skip2PBX, integrated into my Asterisk machine, but it seem > more invasive than Gizmo5 opensky. Doesn't it? Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning there's no software to install on your system. In minutes the system can be working for your Asterisk box. This is like using

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 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

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

Hi, I've spent the best part of a sunny afternoon trying to get Skype to work on my CentOS 5 desktop. My soundcard seems configured OK. I can play sounds and hear them OK in my headset: $ aplay /usr/share/sounds/alsa/*.wav I can record speech from my microphone OK: $ arecord > ~/test.wav ... although I have to say that sound quality seems rather poor. It's quite fuzzy. Other

Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny.

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

By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it.

client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo < rodolfo.alcazar at padep.org.bo> wrote: > Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. > Could be what you want. > > Greets! > > Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: > > TianLun, > > > > I

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

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

Hi All, I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use XML provisioning (or any kind of remote provisioning). I figured since Sipura/Cisco won't release the utility

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 ....