3 Steps to VoIP Nirvana: It’s Incredible PBX 2.0

We’re pleased to introduce the latest and greatest Incredible PBX with an incomparable VoIP feature set. For the Pioneers, you now get transparent support for both Asterisk® 1.8 with PIAF-Purple and Asterisk 10 with PIAF-Red. Just download the PIAF 1.7.5.6.3 ISO and pick your favorite flavor

via Nerd Vittles » 3 Steps to VoIP Nirvana: It’s Incredible PBX 2.0.

Mobile Grabs $813.6M in February Funding

TruPhone: This London-based company develops low cost voice-over-IP software. The company got $118 million in February from the Russian business tycoon Roman Abramovich, who is currently worth over $10 billion, according to Forbes.

via Light Reading – Mobile Grabs $813.6M in February Funding.

From: http://www.truphone.com/en-GB/Business/

Truphone is the mobile phone network specially designed for international businesses.

  •  Save between 30-90% on voice, text and data compared with major UK operators
  •  Enjoy price plans that cover usage in the UK, plus Europe, the US and Australia
  •  Experience business-class coverage in 220 countries
  • Multiple international numbers on a single handset

  • Why do Enterprises Favor Centralized SIP Trunking Topologies?

    Forrester Consulting estimates a 401% ROI for a typical large organization that deploys a centralized SIP trunking topology using Acme Packet E-SBCs. The Total Economic Impact of Acme Packet’s Session Border Controller, which is based on separate and independent research by Forrester, indicates organizations save by eliminating underutilized TDM trunks serving each location as well as lower per-minute rates that apply to VoIP usage. In fact, each of the customers that Forrester interviewed reported a 40%-60% reduction in monthly service fees upon replacing T1/E1 TDM trunk lines with SIP trunks. Obviously, savings is directly proportional to the number of TDM trunks eliminated.

    via Why do Enterprises Favor Centralized SIP Trunking Topologies?.

    VoIPmonitor

    VoIPmonitor is open source network packet sniffer with commercial frontend for SIP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters – delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode speech and play it over the commercial WEB GUI or save it to disk as WAV. Supported codecs are G.711 alaw/ulaw and commercial plugins supports G.722 G.729a G.723 iLBC Speex GSM Silk iSAC. VoIPmonitor is also able to convert T.38 FAX to PDF.

    via VoIPmonitor – VoIP monitoring software – quality analyzer – WAV recorder.

    Free SIP/VoIP client for Android

    For Google™ Voice users, Sipdroid can now create a new, free PBXes account that is automatically linked to an existing Google™ Voice account. The new feature requires Android 2.0, or above, and Google’s app connected to your Voice account.

    via sipdroid – Free SIP/VoIP client for Android – Google Project Hosting.

    Just found this site and sipdroid looks like an interesting VOIP solution for a tablet wifi.  Will download to see how it works.

    Facebook Messenger app change allows free calls via WiFi

    Using software, rather than hardware, the latest update of Facebook’s Messenger app now lets U.S. users place voice calls over WiFi. The rollout follows reports of Facebook testing voice call features in Canada earlier this month.

    via Facebook Messenger app change allows free calls via WiFi – The Washington Post.

    You might want to be careful about this after reading this article on slashdot:

    Facebook Lets You Harvest Account Phone Numbers

    Build a Skype Server for Your Home Phone System

    Configure Linux to Work with Skype

    I’ll assume that because you’re a Linux Journal reader, getting Fedora Core 3 up and running on your Skype server is a no-brainer. The only important thing to remember is that Skype is a Qt application (though it’s also available in a version with Qt statically linked), and the Skype API uses D-BUS. Also, disable the screensaver (after all, there won’t be any screen to “save”) and power standby features as these may interfere with Skype.

    Here’s a step-by-step guide to setting up Linux to work with Skype (it assumes you have set up a Linux user account named skype for the purpose):

    via Build a Skype Server for Your Home Phone System | Linux Journal.