NOTE: The following information comes from http://freepbx.org/news/2009-03-19/restricting-outbound-calls-in-freepbx-whitelist. It’s being copied here in the event the original post becomes unavailable. Please refer to the above web site for current information.
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I just discovered Elastix, It is an ISO image that installs CentOS, Asterisk, FreePBX, and other goodies to create an instant PBX. Compare it to PBX In A Flash and Trixbox.

The Grandstream HandTones have an issue where Asterisk thinks the device is busy after it sits for awhile. Calling the extension programmed for the device will send your call to voicemail. Picking up the handset will result in no dial tone. However if you hangup and pickup the handset again you will get dial tone. Calling the extension will now ring the phone … for awhile, and then the pattern repeats.

Here is a work around for this.
1. Set the Grandstream to NOT register for any extensions (FXS and/or FXO ports)
2. Set the sip.conf entry for the Grandstream to use a ‘host=ip address’ instead of ‘host=dynamic’. (I.e. host=192.168.0.2)

If you are using FreePBX then do not touch the sip.conf file. Instead bring up the extension associated with the Grandstream device and change the “Host” field from dynamic to the IP address of the Grandstream.

Of course you will want to use DHCP reservations for the Grandstream or assign a static IP address so that the IP address does not change.

I can’t explain why Asterisk and Grandstream don’t play nice together when Asterisk is set to use a dynamic address. All that I know is that everything works perfectly when you assign the IP address. Maybe someone else can shed some light on this phenomenon.

I was looking for a small, lite weight software based phone for my Windows laptop that would connect back to my Asterisk PBX using the IAX protocol. iaxLite is perfect and works great. I haven’t tried sipLite (same phone but uses theSIP protocol) yet. Well worth a look if you are using Windows.
Soft Phone : iaxTalk.com !, VOIP::Asterisk::Phone::ATA::USB

iaxLite is a IAX2 softphone. sipLite is a SIP softphone

TheGoldFish.net Blog » Blog Archive » DUNDi Tutorial for Asterisk@Home

Confused by DUNDi? Yeah, so was I. It took me a few weeks of kicking my Asterisk box, but I finally got it working. Here’s a step by step tutorial on how to get DUNDi working with Asterisk@Home. This tutorial was written using an Asterisk@Home 2.5 system with FreePBX 2.1, but should work from Asterisk@Home 2.0 onward.