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Posts Tagged OpenSIPS

ClueCon and 2600hz Project

Posted by Zeeek/Randulo in VoIP

Podcast: Download

Audio : Download file (TS-377614.mp3)

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

Video testing – Next week we’ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.

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OpenSIPS 2.0 with Adrian Georgescu, Bogdan Iancu, Flavio Goncalves

Posted by Zeeek/Randulo in VoIP

Podcast: Download

Audio : Download file (TS-317304.mp3)

An introduction to OpenSIPS 2.0 new design and OpenSIPS Software Foundation. Our guests will be:

Adrian Georgescu (CEO of AG Projects who make Blink)
Bogdan Iancu (CEO of Voice System and founder of OpenSIPS project)
Flavio Goncalves (CEO of V.OFFICE and author of “Building Telephony Systems with OpenSIPS 1.6”

OpenSIPS, its evolution, the new design of 2.0, the effect of new design on the community, availability of more information, the foundation and open source aspect of OpenSIPS.

IRC Links

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Astlinux

Posted by Zeeek/Randulo in VoIP

Podcast: Download

Audio : Download file (TS-306309.mp3)

astlinuxlogo

AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal for embedded and commerical Asterisk based solutions.

On the call Friday we will discuss the newly released 0.7.0 version of AstLinux and the number of improvements over the 0.6 series of releases.  0.7.0 will be available with either Asterisk 1.4.29 or 1.6.21 (two separate download images).

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