Dialing By SIP URI In Asterisk

polycom_ip650_256Barrett Lyon is an interesting guy. He’s the CTO & Co-founder of BitGravity, a company specializing in streaming media solutions for the web.  Long ago I stumbled upon his blog which was then a very simple site. What caught my attention was a post detailing how to use direct dialing by SIP URI within Asterisk. It was unique because he laid out all the relevant DNS changes necessary, and supported the post with a video clip showing the process.

That old blog post has long vanished in a redesign of his site, but not long ago he replaced it with a new version called simply P2P SIP URI Dialing. It’s a good reference on the subject. Highly recommended.

Of course, dialing by SIP URI is really about wholly bypassing legacy telcos and what remains of their TDM networks. It assures that you have IP connectivity between end-points. That simple fact is what opens up the opportunity for wideband calling (HDVoice) as well as video calling and even encrypted calling.

Logitech’s New Squeezebox Touch

If you’ve been around here a bit you’ll know that I really like my Squeezebox systems. In fact, enough that we have four of them and need one or two more to complete my master plan for whole-house, yard & office audio. This past week Logitech announced a new version that revisits the core functionality of the Squeezebox3, the model that I own already.

squeezebox-touch

It’s called the Squeezebox touch and it’s very similar to my beloved Squeezebox3. For the same $299 selling price as the older Squeezebox3 the Squeezebox Touch adds a 4.3″ touchscreen LCD, a USB port and SD card slot for playing music from local media even without a Squeeze Center server.

This makes it just about perfect for students who don’t want to keep a computer running as the repository of their music library. Just plug a portable HD into the Squeezebox Touch and use the infra-red remote control as your means of accessing its functions. Since the web interface was based on the Squeeze Center server software I presume that it won’t be available if you use a PC based music server.

I can see one of these in my future. Perhaps if I’m very, very good Stella will get me one for Christmas?

No Jitter’s Matt Brunk: SIP Means Change

nortel_lo_ip_phone_1120e copyEarlier this week Matt Brunk penned a post over at CMP’s No Jitter blog entitled “SIP Means Change.” It’s a short piece detailing the contrast between SIP phones, Asterisk servers and legacy proprietary equipment. In particular it dwells on the boot times of the various items. He highlights how anything that takes longer, even just a little longer, ultimately has a higher cost.

Matt points out that the older, proprietary digital phones were effectively instant-on devices compared to SIP phones. This is a little obtuse in that SIP isn’t really the culprit. It’s just a protocol. Cisco phones running SCCP would have similar boot times to SIP handsets. I presume that Nortel phones running UNISTIM would also have similar boot times.

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The Gigaset A580IP, Gizmo5 & Google Voice

siemens-a580ip-160This is just too handy to remain hidden away in a comment so I thought I’d make it a proper post on it’s own. Gigaset Communication‘s Tony Stankus responded to a question asked in a comment on my A580IP review. The question was, how to setup the A580IP for use with Gizmo5 and Google Voice?

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Codecs, Wideband & Stereo: A Conversation At AMOOCON

AMOOCON-200Back in the spring of this year VUC founder Randy Resnick made the trek to Germany to attend AMOOCON, a conference for Asterisk Users. AMOOCON started out in 2007 known as AsteriskTAG which literally translated means “Asterisk Day.”

While at AMOOCON Randy made a point of recording some of the presentations. These recordings can be found on the VoIP Users Conference site. Various sessions were recorded by the conference organizers and are available on the AMOOCON site. There they also have quite a lot of video of the formal sessions.

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