AstLinux Users Guide Chapter 1
Posted: August 28, 2006 Return to the AstLinux User Guide Contents 20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ Chapter 1 – Installation Page Contents Choosing…
Read More »AstLinux Users Guide Chapter 0
Posted: August 28, 2006 20 January 2010 Astlinux has now reached version 0.7. These voip-info.org pages are therefore extremely out-of-date. Please refer to the official Astlinux site: https://www.astlinux-project.org/ About the Project Return to the AstLinux User Guide Contents Page Contents How it…
Read More »PJSUA
Posted: August 26, 2006 PJSUA is a command line SIP user agent (UA) written with PJSIP Open source SIP stack. While it is used mainly as the reference implementation of PJSIP, it is quite useful for testing or troubleshooting SIP installations, because…
Read More »PJMEDIA
Posted: August 26, 2006 PJMEDIA is an Open Source media stack written in C and is optimized for multimedia over IP communications. Among the features: it is optimized for small footprint, very very portable (Win32, Windows Mobile, Linux, *nix, MacOS X, Symbian,…
Read More »PJSIP
Posted: August 26, 2006 PJSIP is an Open Source Embedded SIP protocol stack written in C. The development of PJSIP is mainly focused on having a small footprint, modular, and very portable SIP stack for embedded development purposes (although it’s perfectly good…
Read More »Capa Tres Soluciones Tecnologicas S.L.
Posted: August 24, 2006 Capa Tres Soluciones Tecnologicas S.L.CIF- B63846984Telef./Fax 935605424https://www.capatres.com//a> Capa Tres Soluciones Tecnológicas es una empresa joven y dinámica, dedicada a la Consultoría Informática, distribución de material de VoIP y a la prestación de Servicios a empresas. Capa Tres Soluciones…
Read More »Standalone Cisco 7941/7961 without a local PBX
Posted: August 24, 2006 This subject would not be worthy of its own page for most phones, but there are enough caveats and workarounds for these UAs that others may benefit from shared experience. Note that US part numbers are used throughout…
Read More »FreeSwitch Javascript File hasRandomAccess
Posted: August 22, 2006 PLEASE VISIT http://wiki.freeswitch.org for more up to date info. What is it? FreeSWITCH™ is an open source communications platform. FreeSWITCH™ is a library which ships with a small executable that loads the library, launches the core, and performs…
Read More »Siemens Gigaset C450IP
Posted: August 21, 2006 SIEMENS Gigaset C450 IP This is the first mass market DECT / SIP phone. It started shipment in mainland Europe during summer 2006. It sells over the internet for less than €85 ( £58, $109) including shipment. Note…
Read More »avodaq AG – Just Communicate
Posted: August 21, 2006 Expect real specialists and true dedication. Communication technology with all its possibilities is becoming ever more important. Because communication involves more than just telephone, email or fax – and can provide you with important benefits. Take advantage of…
Read More »Virtual Phone Line
Posted: August 20, 2006 Virtual Phone Line is a Enhanced Call forwarding service in service since 1999. Virtual Phone Line won best of the show award for the internet world 2000 show in Los Angles for the technology and service. The company…
Read More »Asynchronous Javascript Asterisk Manager (AJAM)
Posted: August 20, 2006 AJAM is a new technology available in Asterisk 1.4 which allows web browsers or other HTTP enabled applications and web pages to directly access the Asterisk Manager Interface (AMI) via HTTP. Setting up your server to process AJAM…
Read More »FreeSwitch Javascript DB Exec
Posted: August 20, 2006 PLEASE VISIT http://wiki.freeswitch.org for more up to date info. What is it? FreeSWITCH™ is an open source communications platform. FreeSWITCH™ is a library which ships with a small executable that loads the library, launches the core, and performs…
Read More »Uniden UIP1868
Posted: August 19, 2006 Uniden UIP1868 SIP Phone Features Includes both wired handset on base unit and 5.8GHz wireless handset Both have speakerphone SIP Phone with support for two lines Codecs: G711u, G.711a, G.729, G.726 Both lines must use the same server…
Read More »Asterisk gui tycho
Posted: August 18, 2006 What is “Tycho”? Tycho is a java based voicemail client (not only) for asterisk which enables you to manage and listen to your voicemails pretty much like your manage your email. In addition, we have added some dial…
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