After completely rewriting ENUMPlus from the ground up (2 days total) we have come much closer to a universal ENUM lookup source.  The new module can be downloaded here so install it now! Here’s why :

• The module forms a HTTP request to our API which queries many other sources and returns the most accurate address.
• There is no longer a need to add multiple lookup sources to the old enumlookup AGI.
• It audibly notifies you when the call is made over ENUM.
• You can track your calls in the panel.
• White pages / search functionality (including avatars )
• Call validation is much more stable.

If you don’t have an account, get one – registration is quick and simple and you will be a happy geek.

Check out the project page / wiki / site / sourceforge / FreePBX / voip-info / piaf for more info.

Recently we (#pbxinaflash) have been playing around with ENUM at e164.org, some with greater success than others.  We have all experienced intermittent authentication failures and “missed calls” – overall not really impressed with the whole service.  It seems their services have been dwindling quality wise for some time now.

We have begun coding a new project for e164 with security, quality of service and ease of use in mind.  We plan to have immediate call out for registration authentication assuring proof of number and SIP URI.  The project has only been in the works for two days but we have been moving very quickly.

If you are a Bind expert (or just want to help) please join us on freenode at #pbxinaflash and submit your two cents.  Our temporary project page can be found at http://sip.geekhut.org – we would love some feedback.

Thus far we have a solid registration system implemented that does everything but write the DNS records (useless at this point) – that’s why we need you to help with Bind configuration.  This is a Debian (Stable) host running a minimal LAMP stack and chrooted Bin9.

We chose to call it “ENUM Enhanced” as we plan to pass additional contact information (avatar, email, address, trade, alternate contact etc) complete with a screen pop App that will display user specified information upon incoming call.

Once completed we plan to release the source GPL and freely available to everyone – each implementation will phone home and share its public records with our global database (opt-in of course) which should encourage use and expand the userbase exponentially.

We look forward to hearing from you and should have a public SVN up within the week.

I have been a busy beaver today preparing two new modules for Asteristickies,

I recently completed (and uploaded) the latest release featuring notificxations for your XBMC.   It supports authentication as well as specifying a custom port (for computer builds with limited perimssions (ports < 1000))

The script pops a tiny message at the bottom of the screen and pauses any currently playing media.  There no modificatios to the dial plan and no third party software to install.

To get a copy, sign up at the PiaF forums and PM user “aster1sk” with your email address.

I got an email a few days back regarding the Jabber interaction with Asteristickies with some great suggestions for future features :

Hey Greg!

Wow, watching the forum, you’re really going to town on this!  Awesome stuff.
I’ll have to install a newer build!

Perhaps you could think about a dream integration  I have for
Asterisk/Jabber IM…

Imagine this:

- You have a jabber account
- You have a PIAF box.

You build a script that does this:

1 – When an incoming call comes in to a specific DID, you get the
following Jabber text message:

You have an incoming call to NXX-XXX-XXXX from ‘John Doe’
What would you like to do?
1 – let call go to voicemail (default; same as not responding)
2 – transfer to NXX-NXXX (pre-configured number)
3 – transfer to number entered in IM
4 – TTS text you type.

So, for example, if you were sitting in a meeting and an important
call notification came in,
you could, for example, TTS “I’m in a meeting, I’ll call back in 10 minutes”
Or, if you expecting a very important client call, force a transfer to
your cell phone.  A way to
circumvent standard behaviour.

For this to work, it must all happen between the 1st ring and the
timeout to voicemail.
I think this would be a VERY awesome integration… don’t know how to
do it on linux, as I’m a
.Net Windows guy.  I’ve built a mashup on Windows + Linux but just for
concept testing.

So I decided to investigate further – this is what I came up with.

Still only a proof of concept, you can imagine how this could benefit those with mobile devices and support teams.

I have successfully rolled out call transfer and call out, next in line are TTS (almost ready) and call forward.

Joe Roper has written a spectacular article (and script) for those looking to beef up their Asterisk boxes with a powerful fax server. HylaFax is by no means new – but the simplicity of configuration and minimal effort required now to get an enterprise class fax service in minutes is ground breaking.

I just set it up in less than ten minutes and am sending + receiving faxes over VoiP (your mileage may vary) over my IPKall extension! Heres proof :