similar to: Problems for Skype for Asterisk

Is there anything special that has to be done to make video calls work? It doesn't seem to work for me (no video). What CODECS are supported?

>> >> >> 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> >>

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

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, 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

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 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 authentication get access to internal PBX extensions. I CAN do this with a PIN, no sweat, but from a user perspective it

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, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", "computer-temp.sh,darwin,") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame

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

Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel

Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype "user". I do not need to call landlines via Skype. 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and route the call to a specific Asterisk SIP extension. At first, I thought it would be

I am running asterisk 1.6.2.6 and have configured hints for our extensions and have a couple of Aastra 6755i test phones. The phones register fine but 'core show hints' shows the lines as idle even if they are in use. I read the wiki and see mention about needing to set call-limit in asterisk 1.4 but that has been depreciated in 1.6 so what is the way it should be done in 1.6?

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

Which is the best version of skype to install in CentOS 5.3? skype.i586 2.0.0.72-fc5 from the skype repo or skype 2.1.0.47-fc10.i586.rpm from the 'skype for linux beta' downloads section of the skype website? Andy

Earlier last year I had a 5.7 system running with skype. It worked fine. Later last year I had to reinstall my entire system. I didn't install skype then. Yesterday I tried to find a version of skype to use on 5.7, but couldn't find one on skype.com; there's no version for RH/CentOS 5.7. Does anyone know where a working skype can be found? Thanks.

Hey Y'all, About a week ago my Skype installation stopped working. I can start it up but I can't log on. Does this have something to do with the recent update to LibC? Is there a repo som where that has the latest version of Skype for CentOS 6.5? When I go to the Skype web site I see that there is a version several versions newer available for RPM based Linux but that one won't

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

As I have no friends and no life I thought that I would set up my asterisk server with Skype. 1) Paid the $, got the licence, built and installed 2) create a business skype account (called company "foo") 3) created a member of the business called "bar" 4) updated the skype conf file 5) restarted asterisk => skype show settings Skype For Asterisk Settings: