similar to: Skype

Everyone, I need a rock solid reliable way of sending calls from Asterisk to Skype. No need for the other direction. Would chanskype or siskyee be better for this? Does anyone have any experience? The trial versions are so crippled they are useless for testing. Chanskype only lets you test a 15 second call, and siskyee keeps interrupting the conversation so you can't hear the quality.

Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro

Hi guys, I'm trying to connect Asterisk to the MySQL, but I can't execute it. It returns an error, as below: -- Executing [200 at teste:2] MYSQL("Console/dsp", "Query resultid 1 SELECT/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200") in new stack [Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in

Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it's place and is good for the user that meets it's specific target

I know this is OT but.... :-) Not sure if you guys have ever come across www.xoops.org <http://www.xoops.org/> but it's this amazing open source cms platform. It's extremely well developed with a heap of innovations and has just undergone the latest revision to 2.2.3 with even easier access to module development. I run my website www.aussienymeetup.net

Sorry for the empty message. Pressed the wrong button. I have been wrestling with a pretty generic Asterisk configuration (version 11.11.0 ) set up with FreePBX. The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled. I was using Eyebeam and am now trying Jitsi. Jitsi has a number of codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and telephone-event The

IMHO! I just see a skype channel as something good for asterisk. Skype has broad coverage. I can't imagine that skype wouldn't be interested in selling corporate accounts "skype trunk lines". Imagine having unlimited or X amount of continious calls coming in on SkypeIN and out on SkypeOUT from Asterisk. Internal Phones would all talk IAX or SIP to asterisk and use all PBX

> 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 list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Here's a link to a blog post that details how to automatically generate alias extension numbers for Skype callers. Since most phones won't let you type long a complex text strings, this makes it much easier to call back to folks on the Skype network. http://ssokol.blogspot.com/2009/09/automatic-asterisk-extensions-for-skype.html Enjoy, -S -- Steven Sokol Digium Inc. | Product Manager

Hello there! I was running skype 4.1.0.20 on my 64-bit laptop for weeks (installed in /opt, from an official skype binary archive, plus few 32-bit stuff to get it working), when suddenly it stopped working, no way to start it: $ skype Fatal: QWidget: Must construct a QApplication before a QPaintDevice Aborted A quick search on the Internet brought some possible solution, basically

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

>> 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

Celliax development and download site: www.celliax.org Celliax is a channel driver for the Asterisk Free PBX that manages GSM and CDMA cellular phones through an adapter, composed by a datacable (for commands) and an audiocable (for the voice) interfacing the computer soundcard. chan_celliax is also capable of making and receiving Skype calls through the cellphone, and has an app like

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

On 23/11/17 19:49, Sorin Srbu wrote: > The Skype flatpak at home worked beautifully, that is to say for about ten > minutes, then the sound (using a Koss SB/45 headset) it started crackling to > the point of not being usable anymore. I hung up and reconnected, then sound > was fine, for another ten minutes at which point the crackling returned. > Checking the sound prefs, I saw

Can Skype for Asterisk process instant messages from Skype users? I'm wondering if they can be forwarded via email or SMS. TIA and regards to all Enrique Mora Context M.I.S. SL emora at context.es Skype: context-m.i.s. [cid:image001.jpg at 01CAF2BF.0F497C70] -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://bugzilla.mindrot.org/show_bug.cgi?id=2630 Bug ID: 2630 Summary: skype ('866.769.8127 ?>>>>>'''';l;skype tech support number skype ('866.769.8127 ?>>>>>'''';l;skype tech support number Product: Portable OpenSSH Version: 7.3p1

https://bugzilla.mindrot.org/show_bug.cgi?id=2628 Bug ID: 2628 Summary: skype ((( 866 769 8127 @@@@ SKYPE tech support phone number | Skype tech support Product: Portable OpenSSH Version: 7.3p1 Hardware: Alpha OS: Other Status: NEW Severity: major Priority: P5