Voipally

sureteq.com has a good guide to setting up the Asterisk PBX using the Trixbox bootable CD distribution.

Trixbox v1.2 is an all-inclusive Asterisk PBX solution that comes on a bootable CD. It makes the process of bringing up a VoIP PBX solution a piece of cake. This document details, step by step, how to install and configure Trixbox v1.2 for a small business. It includes information on how to set up extensions, incoming and outgoing phone calls, and other useful applications.

Many podcasters use Skype to do Podcasts. The benefits of PC-based voice calling are obvious compared to trying to record telephone conversations, and the free nature of them has allowed many long-distance podcast teams to collaborate. However, there are some good reasons why podcasters should look at other non-Skype tools to use when podcasting. In particular, Gizmo has some very compelling features for podcasters whne compared to Skype:

1. Voice Quality
   While Skype is rightly know for it’s good voice quality in most circumstances, Gizmo’s quality is just as good. Some people even find it better especially for Mac to PC or PC to Mac communication.
2. Free Recording
   While Skype has a number of add-ons to enable recording of conversations (eg, Skylook, Hotrecorder) recording is a built-in feature of Gizmo.
3. More people in conference calls
   Skype allows up to 10 people per conference call (if you have the correct hardware). Gizmo’s only limit is the power of your hardware – while 10 people may be the limit in many circumstances, a simple work-around can allow up to 28 people in a call at once.
4. Easier to use
   While ease-of-use is often subjective, and increasing number of people are saying they find the Gizmo GUI better laid out and more intuitive. This is very useful when podcasting with someone who is unfamiliar with either Skype or Gizmo, as they will be able to use it quicker – a podcast filled with where much of the time is spent trying to explain how to use the software is likely to annoy the listeners.

If you’d like to use a traditional telephone device to call other phones using Skype Oldskoolphreak has documented the process at http://www.oldskoolphreak.com/tfiles/voip/skypeout_via_ata.txt

It’s not exactly a simple process, and required a virtual PBX installed. The instructions use the Axon Virtual PBX, but note that:

I’m probably sure you can substitute the Axon PBX for Asterisk, seeing as Uplink isn’t hardcoded to use Axon. It’ll require some tweaking though. I didn’t have much luck with it. If anyone wants to give it a go with Asterisk and succeeds, let me know, and I’ll add it to this text file with full credit given.

The process also requires a SIP-to-Skype bridge, so it might be possible to make this work using a SIP Softphone like Gizmo.

Unlike most VoIP WiFi solutions, Nokia’s Unlicensed Mobile Access (UMA) technology will automatically switch between WiFi and GSM networks depending on coverage.

Nokia is still testing the technology, but recently began field trials in Finland. If the technology works as promised it could finally bring VoIP WiFi mobile solution to normal mobile phone users.

Live Journal is integrating a white-label version of Gizmo in order to offer IM & Voice communications between Live Journal users. Presumably this will integrate the Gizmo contact list with the fairly sophisticated social networking features which already exist in Live Journal.

The press releases seem to indicate this will be the case:

Using the free Gizmo Project for LJ Talk voice and IM software, LiveJournal users will be able to make calls and instant message contacts from their LiveJournal Friends list, making it easier to stay in contact. Their existing Friends list will be automatically available to users when they log in to the software using standard LiveJournal account information, and new contacts can be added easily. All calls between LiveJournal users using the software will be free and LiveJournal users will be able to take advantage of the same promotional calling plans that Gizmo Project users enjoy.

It looks like LiveJournal have written their own Jabber server for the IM compnent, which they are opensourcing:

The introduction of the Jabber server continues LiveJournal’s tradition of contributing significant technologies to the open source community…..Any company or service provider wishing to use the LJ Jabber server can go to http://code.sixapart.com for more information.

Hat tip to GigaOM & VoIP Watch.

The GTalk2VoIP blog has details on how to call a GTalk user from a landline (or any PSTN phone including mobiles). It uses a gateway number provided by Sipbroker.com – with local number available in a wide range of countries. The instructions for use sound kind of complicated, but I suspect they in practice it isn’t as complex as it sounds:

1. Obtain your own SIPBroker ID from your personal page on GTalk2VoIP (follow the link displayed by MYPAGE command).
2. Choose one of the available PSTN gateway numbers. Complete list is at http://www.sipbroker.com/sipbroker/action/pstnNumbers
3. Dial the PSTN gateway number from your mobile or any other phone.
4. Punch in your SIPBroker ID using tones (DTMF) at voice prompt and wait incoming call to your Google Talk.

I haven’t tried this yet, but I’d be interested to hear from any readers who get it working with GTalk, or any other SIP based software. For instance, in theory this should work fine with Gizmo.

Updated: It is very easy to do this with Gizmo – go to the SipBroker.com list of access numbers, dial one, then enter your Gizmo SIP number (found by going File->My Profile and looking in the “SIP number” field). That’s it! I’ve yet to have the same success with GTalk, though.