OnSIP & Android 2.3 (a.k.a Gingerbread)

Earlier this week the Android development team released some news about Android v2.3, aka Gingerbread. This has a lot of people more than a bit excited as one of the new features is a built-in SIP stack.

Some point to this as being a major step toward the integration of Google Voice and Android devices. With a SIP stack included in the OS it will be much easier for people to develop soft phones that leverage the hardware and the data side of Android handsets.

The built-in SIP stack will work with any SIP service provider, giving users tremendous flexibility in choosing who gets their business. It also means that close integration with an IP-PBX will be easier than ever before.

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How To: HDVoice In PBX-In-A-Flash

I recently received an email from someone asking about enabling HDVoice in PBX-In-A-Flash.

I’m about to implement a PBXact for our small company, and I have a nicely running PBX in a Flash in my home. I’m wondering if I can HD-Audio-ize the home rig.

Here’s why I ask. We currently run a SIPX PBX in my company. Everything in that box, all the sound files, are all HD. When you talk to it with an HD endpoint, everything just talks HD. It’s a no brainer.

In my home PIAF/FreePBX/Asterisk, nothing is HD. How do I go about HD-izing it? I have the G.722 codecs all turned on, but I’m wondering if there is an easy way to make all HD-capable endpoints automatically talk HD to each other, and to talk HD to the PBX itself.

Any help is appreciated!

Thanks,

Joe
Fort Collins, Colorado

Since PBX-In-A-Flash is built upon Asterisk there’s a a good chance that this is possible, but its way outside of my scope. Given my employers migration to the OnSIP Hosted PBX I haven’t run Asterisk seriously for some time.

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USB & Bluetooth In IP Phones: A Missed Opportunity?

Last week’s VUC call with FWDs Dan Behringer brings to mind a common complaint about SIP desk phones, namely the lack of an alphanumeric keyboard. Lacking a proper keyboard it’s difficult to really push the idea of SIP URIs as a primary means of making calls.

There are a variety of approaches to overcoming this, including the use of ISNs as prescribed by the Freenum project. That project proposes a means of dialing SIP URIs indirectly, assigning them ISN numbers. Since ISNs use only numbers and the * key they can be dialed on a traditional phone keypad. It’s essentially a way of avoiding SIP URIs through indirection.

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Nerd Vittles Guide to ISN

Aastra_9480i_bl_270x270Over at Nerd Vittles that Asterisk Guru Ward Mundy has posted a great how-to implementing ISN lookups with his popular PBX-In-A-Flash Asterisk distributions. It’s a great overview of ISN and ITADs, including Freenum.org.

ISNs are a novel approach to bridging the gulf between the worlds of legacy phones with traditional phone numbers, and the realm of SIP URIs. It’s a way of dialing something like a normal number and having it automatically be resolved to a SIP URI. This is easily implemented using Asterisk, especially when Nerd Uno shows you how.

All of this is an approach to transitioning from our current state into a time when SIP URIs are more commonplace. That would imply that more of what we think of as “phone calls” would transit a pure IP network. That fact is a major enabling mechanism for such improvements as wideband audio & improved call security, amongst many other things.