Trixbox Business Phone Solutions Welcome. If you've found this page, you were probably expecting to visit a website all about trixbox. You have. (Sort of.) Trixbox was the name of a product offered by Fonality. We had a bunch of product names and it was confusing, so we've consolidated all of our products under the Fonality brand.
OpenWengo, le Wengo open-source INFORMATION : Si vous souhaitez la survie d'un web libre et gratuit, merci de désactiver votre bloqueur de publicité ou de nous mettre en liste blanche. Pour rappel, la publicité diffusée sur ce site est notre unique source de revenus permettant de vous proposer gratuitement ce contenu en finançant le salaire des journalistes. Plus d'explications Skype racheté par eBay, ça a de quoi faire réfléchir. Eh bien 9Telecom a réfléchi.
Астериск и Н.323 Трудной темой для новичков в Астериск является подключения по протоколу Н.323. Напомню, что популярными драйверами каналов являются chan_h323 компании NuFone, его исходники заложены в исходниках Астериска в директории channels, и chan_ooh323 компании Objective Systems. Он идёт в составе Asterisk-addons. Первый труден в компилляции, требует внешних библиотек openh323 & pwlib и компиллятора именно тех версий как указано в README - Open H.323 version v1.18.0, PWLib v1.10.0 and GCC v3.2.2.
Gamification We throw around the term gamification, but what does that really mean? Gamification was originally inspired by the game design industry. Originally, game mechanics were applied to various types of digital experiences to make them more fun and engaging. While there were successes, it became clear that gamification alone didn’t retain an audience over time.
Real-Time Communications with Tropo and Node.js The other day, Jason Goecke wrote an awesome post about using Tropo with WebSockets. I was inspired by Jason’s work to try and replicate it with a slightly different kind of technology. I also wanted to use Node.js. Make Communicating Seamless MIB Parser Configuring NAT traversal using Kamailio 3.1 and the Rtpproxy server This article continues on series of articles about the Kamailio 3.1.x SIP proxy server deployed on the debian lenny and its features. In previous articles we have focused on: 1) installing clear Kamailio 3.1.x server Telephony Black Magic with Tropo, Node.js and Redis In a previous post and screencast on this blog, I demonstrated an example application that highlights some of the more unique features of Tropo to support realtime applications. Tropo’s unique ability to write to, and read from persistent socket connections during the execution of a call sets it apart from other platforms, and creates opportunities to do things that other platforms can only dream of. In this screencast, I build upon and extend my earlier example using Redis, Node.js and jQuery to allow a user to inject text into a running Tropo session and have it read out over the phone using Text-to-Speech. Here is an overview of the technical components of this example application (opens in new window). What I get really excited about in this demo application – the code for which can be found on GitHub – is the ability to send TTS output to multiple subscribers. How cool is that?!?
[#FS-689] No NOTIFY MWI when registering via proxy - FreeSWITCH Jira If a UA registers with FS via proxy, not NOTIFY MWI is sent back to it (I suspect this is not confined to MWI, I believe any NOTIFY will not reach the client). Setting NTA_DEBUG=9 and NUA_DEBUG=9 shows that nta is complaining about the format of an uri passed to it: nua(0x97e8480): adding notify usage with event message-summary nta_leg_tcreate(0xb7944138) nta outgoing create: invalid URI nta: outgoing_free(0xb7971470) nua(0x97e8480): event r_notify 900 Internal error at nua_client.c:711 nua(0x97e8480): removing notify usage with event message-summary
20 FREE SIP Softphones I occasionally run into folks who are looking to deploy softphones versus traditional, desktop-based IP hard phones….and am often asked what softphone technologies are out there that are compatible with SIP based IP PBX platforms such as Asterisk and Trixbox. Below is list of the more popular SIP softphones, all of which are completely free to use. QuteCom Previously known as WengoPhone, Qutecom is a free, SIP compatible VoIP softphone initially developed by Wengo. QuteCom supports a range of VoIP codecs including G.729, G.711, iLBC, G.722 (wideband) and Speex. H.263 for video is also supported. XLite from Counterpath A very popular, free SIP softphone supporting a range of codecs and also offering great support for desktop business video conferencing.