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Hello This UPS is compatible ? http://www.voz-ip.com/apcbackupscs500-p-223.html Thanks -- Mikel Jimenez Fernandez Irontec, Internet y Sistemas sobre GNU/LinuX - http://www.irontec.com +34 94.404.81.82

...es several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) * Fixes thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) * Be more tolerant of what URI we accept for call completion PUBLISH requests. (Closes issue #18946. R...

...es several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) * Fixes thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) * Be more tolerant of what URI we accept for call completion PUBLISH requests. (Closes issue #18946. R...

...1.8+. (Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj) Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4 Information about the security releases are available at: http://downloads.asteris...

...1.8+. (Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by alecdavid, Irontec, ZX81, cmaj) Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4 Information about the security releases are available at: http://downloads.asteris...

Hi all! I've been reading this list for a few weeks and now this is my first post. :-) I'm planning to build a new VoIP telephone system at our company. It's just a small company with not more than 3-4 employees. The telephone system is not so important for us because each employee has it's own mobile phone. Because our company is a small one, we don't want to/we

Hello I am making backup of a Plesk Debian server to /backup using Rsync. My questioin is how can I preserve the ctime, mtime, and atime of original files? Thanks