similar to: Skype for SIP

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/ 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

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

-----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

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

Dear useRs, I would like to query an Access database through R based on a date attribute but, unfortunately, I fail to do so. For example, the table test_table of the test.mdb looks like: ID cd market competitor dd price 1 1 2007-11-20 atl-bos delta 2007-11-20 210 2 2 2007-11-20 atl-bos delta 2007-11-21 190 3 3 2007-11-20 atl-bos delta 2007-11-22

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.

>> >> >> Hi All, >> >> After getting licences for Skype for asterisk a while ago I finally got >> around to setting up a server with two channels and setting up a bcp on >> the skype end. >> >> The idea behind this is the following: >> >> Users can dial into the PBX, get authenticated and only after> >>

It seems that Access needs that you surround the dates with a # symbol. You probably need something like. res <- sqlQuery(channel, "select * from test_table where market = 'atl-bos' and competitor = 'delta' and dd = #2007-11-20#") Hope this helps, Stefano -----Messaggio originale----- Da: r-help-bounces at r-project.org [mailto:r-help-bounces at r-project.org]Per

<!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

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

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

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

Hello, I'm trying install Samba 4 by this link http://wiki.samba.org/index.php/Samba4/HOWTO . The step 2 compile and step 3 install completed, but step 4 provisionhas following error: [root@samba4 setup]# ./provision --realm=sfa.local --domain=sfa --adminpass=5555 --server-role='domain controller'Traceback (most recent call last): File "./provision", line 32, in ?

Hi, I am trying to find out the scale efficiency and optimal scale of banks by stochastic frontier analysis given the panel data of bank. I am free to choose any model of stochastic frontier analysis. The only approach I know to work with R is to estimate a translog production function by sfa or other related function in frontier package, and then use the Ray 1998 formula to find the scale

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

Hi, after having tested SFA in august, I didn't use it for some times and now I receive the subject error when calling through Skype channel. Has anyone an idea on what can be the problem? Thanks -- Daniel

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm