similar to: SkypeKit

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

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

I am using Ruby Feedparser and keep running into situations when a feed is valid but feedparser throws an exception - REXML::ParseException: Declarations can only occur in the doctype declaration. The following feed does validate at feedvalidator.org (with a warning, but does validate): http://www.engadget.com/rss.xml However, when I run feedparser I get this error: <REXML::ParseException:

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/health

On Sat, Oct 17, 2020 at 3:38 PM Robert Heller <heller at deepsoft.com> wrote: > I did a yum update on my CentOS 6 laptop and it upgraded Firefox to 78, > and > FF stopped seeing my mic and speakers. Is there some magic I need to do? > As > a short term (?) fix, I downgraded back to Firefox 68. My system is > otherwise > up-to-date. > Are you sure it's not the

hello, i have a worms armageddon cd iso (made with k3b), now i want to mount that for wine, because i cant play a fast game when i allways have to change my cd. (there r no copyprotection problems). what i have tried: simply mount the cd to /media/WA then i settet in winecfg this folder to be my cd. failled and i tried doing like this: ln -s wa.iso ~/.wine/dosdevices/d:: but the result in

Hiya, SIP Messaging is implemented in asterisk-10... >The only documentation I can find talks about a patch and is pretty old:>http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging <http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging> Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs.... Regards

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

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

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 --------------

Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something like chan_datacard... Regards, Jay -------------- next part -------------- An HTML

Here''s another product which has removed the hassle out of disk management: <http://www.drobo.com/products_demo.aspx> I wonder if they (Data Robotics) will make the Drobo work with ZFS once Leopard is out (since it supports HFS+)? ---8<--- Data Robotics has just introduced Drobo, the world?s first storage robot. Drobo is a direct attached storage array that provides

-----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

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hi all,<br> I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version).<br> Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk

Hi, over the 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

Hi, > The flexibility of this does, though, encourage stuff like 96bit audio. > Anyone implementing a codec which uses this, and import/exports it, will > also write the appropriate conversion OggStream plugin which will allow > applications which only support, say, 16bit audio, to work with it. Do you think the noise in your 16bit application will sound different between a

i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another 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

Hi All, I would like to ask for some advice how to solve following situation: We have to record and decode isdn PROGRESS message and when particular message is found call should be hang up and dialplan should continue. So far we have come up with two ways we think solve the problem and would very much appreciate to hear your opinion. 1) add special progress extension to dialplan so we can

On 08/27/15 12:12, Alice Wonder wrote: > On 08/27/2015 08:58 AM, g wrote: >> On 08/27/15 09:31, m.roth at 5-cent.us wrote: >> <> >> >>> *sigh* >>> And they're probably sent by a script running on the PC of a fat, >>> 47 yr old guy living in a basement and making money this way.... >>> >> . >> i seriously doubt it.

Hello List, i have a spelling and reading software i would like to get running for a primary school because they should not be using windows and the teachers really want that software. I have already tried to get it running with a few people in #winehq, but it seems that its more a guessing game than a streight debugging and solving way. What steps do i need to do to get it running? First of