similar to: Gizmo SIP / Skype gateway

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

> 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 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

>> 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.

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

I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence is username at opensky.gizmo5.com but that gets mapped to sipphone address so I set that up to map

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, 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

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

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 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 ....

This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI> module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message

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

Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? Thanks --------------

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 found this today, I am not a skype user but have read on chan_skype and don't like aspects of how it is implemented. My thoughts on it are only theoretical as I haven't used it I just cringe at adding X to a server. Anyhow there is a new project called sippyskype that appears to do a similar sort of thing with a couple differences. 1. Its

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension