search for: chan_skype

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 [Sep 15 11:08...

Hi guys, is there a comment from digium on the license of chan_skype? I could not find the GPL_KEY in the precompiled module, and they don't release the source. So i'm guessing, they'd need a commercial license...? Regards, Andreas _________________________________________________________________ Become a fitness fanatic @ http://xtramsn.co.nz/heal...

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

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm

-----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 FREE (as in beer)...

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/...

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 d...

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 -------------- next part -...

...anager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found<br> [Dec 28 15:18:12] VERBOSE[2436] logger.c:   == Manager 'jabber' logged on from 127.0.0.1<br> [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start.<br> [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library.<br> <br> Skyhost it seems to be expired and then "skypeforasterisk" and "skypewatcher" doesn't start at all.<br> <br> Anyone have some information about SkyHost expiration ?<br> <br> Thanks,&l...

...other skype user i Succeed to send only using for example: exten => 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(<account>,<from>,<timeout>),and where and how I should add this code in extensions.conf my chan_Skype.conf [Account] secret=XXXXXX context=from-pstn exten= Account disallow=all allow=g729 allow=alaw allow=slin allow=ulaw auth_policy=accept buddy_presence=yes direction=both ;auth_policy=ignore buddy_autoadd=true ;buddy_presence=no mohinterpret=default ;mohsuggest=none Regard...

...e year, the "IM" question popped up here and then, the answer was allways something along the line "yeah, there's an experimental patch, but it's not yet ready"... Are there any plans to change this, to ie. allow messages between xmpp and sip or iax? Especially with the chan_skype (that's hopefully is going to be released in my lifetime ;)) a messaging infrastructure in the core would be interesting... Regards, Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090729/c4dae...

http://www.engadget.com/2010/06/22/skypekit-beta-sdk-adds-skype-to-any-application-or-device/ Great! Finally a change to get a chan_skype without beeing a**-raped by the copyprotection (which is the sole reason i didn't buy it), and maybe even more than the absolute basic features (like Silk and video)... J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk...

Good Evening, I'm setting up an Internet Radio website with call-in functionality, and need to know the kinds of FOSS tools I should install to get the job done. Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png Call protocol: [Producer calls in] [Host calls in] [Guest calls in]->[Screened by Producer, if accepted, conferenced into host] On the website

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

Just got a call from Ebay's unwired buyer and "The Voice" is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). Thanks, Steve

Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +612421100000 to something like 024221100000 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XXXXXXXXXX but that still seems wrong to me. it could match other numbers.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers...

I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium?s web store and purchase (for zero dollars) the

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

...yes > > They demoed it - everyone seems pretty confident it works > as advertized. > No wide-band codec (yet) > > Tim. > > On 25 Sep 2008, at 17:55, randulo wrote: > > > I know a lot of linux and open source people think it's superfluous, > > but a pseudo chan_skype is huge (assuming it works as advertised). It > > means anyone with Skype can connect to your server presence. And > > presumably you can call people via Skype. And use Skype out, etc. > > > > > > > > On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness > > <st...