A few days ago on Digium’s blog Malcolm Davenport posted some initial details about the companies new line of SIP phones. Malcolm’s post presents our first formal glimpse into the relationship between the phones and a new module for Asterisk called the Digium Phone Module For Asterisk (aka DPMA.)
You may recall that various folks from Digium were on a VUC call back on Feb 3rd, not long after the phones were announced and shown at IT Expo.
While I am not currently an Asterisk user I still try to stay in touch with what goes on in that realm. Earlier this week Rod Montgomery penned a post at the Digium blog entitled “Top 10 Tricks You Didn’t Know That Asterisk Could Do.”
The post is structured like a Top 10 list. Most of the items included are genuinely useful. However, right at the bottom in the #1 position, he highlights newfound support for very high-sample rate audio, aka Super-Wideband.
As you likely heard on April 27th Northern Alabama suffered a spate of violent storms, including a number of large tornados. Many thousands of people were impacted, including long term loss of power and network connectivity. Digium was amongst the many, many businesses impacted by the events of the day.
I must commend Danny Windam, CEO of Digium, for making excellent use of the corporate blog to keep the Asterisk community informed about the companies operational status. Some time having elapsed since the unfortunate events of that day Digium has since returned to normal operations.
Today marks the launch of the Freetalk Connect PBX from In-Store Solutions. Freetalk Connect is a small business PBX appliance that includes close integration with Skype.
Freetalk Connect was initially announced in January at ITEXPO East. Thomas Howe, CTO of In-Store Solutions, made a guest appearance on a VUC call on August 13th to discuss the device. At that time we learned that Freetalk Connect is based upon Asterisk and Skype-For-Asterisk.
Digium's Tristan Barnum, Director of Product Marketing for Switchvox, has a nice piece over at IT Business Edge called , "Five Things You Didn't Know About VoIP" I especially like item #5 on her list, "SIP Trunks don’t exist. There’s…
Having read & listened this far into this series you should now have some grasp of how narrowband (G.711) compares to wideband (G.722/G.722.1) and even super-wideband (G.722.1C) audio for telephony applications. The differences in many cases are quite pronounced, even startling. What you hear in the examples are just the most obvious properties of the encoding, sampling rate and by implication, the available audio bandwidth. It’s worth understanding a bit more about the evolution of the role of the codec over time. This will help you frame up how the Siren codecs fit into the Asterisk realm.