search for: qutecom

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for CentOS 5? Thanks, -Bo...

Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,grandstream,aastra, qutecom, eyebeam, ...) digium need feedback for srtp inclusion to 1.6.3.0 http://bugs.digium.com/view.php?id=5413 if you need additional info, i'm on jabber - cervajs at njs.netlab.cz thanks! --------------------------------------- Marek Cervenka ======================================= ___________...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this problem. Can it be due to a configuration problem of the Grandstream phone? Th...

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. is it possible to run system like that ? is it good idea , can

...lt;asterisk-users at lists.digium.com> Message-ID: <alpine.LRH.2.00.0905210116570.17084 at axpsu.fpf.slu.cz> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > can anybody help me to give Opensource SIP client information which can be modified as per our requirment http://www.qutecom.org --------------------------------------- Marek Cervenka ======================================= ------------------------------ Message: 8 Date: Wed, 20 May 2009 16:33:15 -0700 From: Jonathan Thurman <jthurman42 at gmail.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and Mee...