I have received 6 calls over the past month from readers who are having difficulty provisioning their VoIP hardware (specifically Cisco endpoints) and in most cases was able to help get the devices registered – mind you it was quite a hassle over the phone.  The biggest problem is that for the non-geeks this can be a daunting task and many don’t care to spend the time.

The good news is that we have gathered a team of developers to write a totally web based app for provisioning your endpoints with ease.  To start we will support the devices that we actually have and can test.  Within the week we plan to support the following models :

• Grandstream GXP XXXX & Budgetone Models
• Cisco 79XX series
• Several Aastra models (53i, 9143i, 480i)
• Linksys SPA94X
• Sipura SPA2102 (AKA Linksys PAP2)
• Whatever else can be found or generously donated.

We began development on the frontend which has two modes – wizard (basic) and advanced.  We have also constructed working templates for many models listed above which is probably the most tedious task.  We will offer HTTP and TFTP provisioning totally free! as well as the option to download your entire endpoint configuration to place on your own network (if you so choose).

A word on security :

Any and all data you submit to this application will be passed through an extensive encryption / decryption process and stored in the database – thus no one but you will have access to the private information.  You will actually have the ability to set your own security salt and change it if you suspect foul play (which is very unlikely).  We may also buy a CA for that extra peace of mind.

We will be adding this functionality as a module at ENUMPlus so sign up to catch first glimpse.

Anyone looking to help either with a donation of hardware or Paypal is of course much appreciated.  The more devices we can get our hot little hands on – the more we will be able to support.

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.

Well I just received an invite to Google Voice and I must say I was quite upset to discover that it is not yet available in Canada… nonetheless I had a socks box in Cali so I tunnelled my registration and was ready to rock.

For those of you unfamilliar with Socks – it basically pushes all requests through a secure connection on a remote system and is quite simple to do – take a look at my “How to Watch American TV in Canada” post from a while back.

Once I was in I ran into another little snag, you can not set your call-in number to an international DID (in my case a Canadian DID).  So I just pointed them to my free IPKall number and was really cooking with gas (you too can get a free IPKall DID at ipkall.com).

For those of you anticipating an invite yuo can expect the usual Google UI (thank god (I adore simplicity)), you can also look forward to transcribing your voicemail messages to text – and have them emailed.  This functionality has been lacking in Asterisk which is why I believe this is going to drop a nuke on the Asterisk devs to get this module out – so support your * developers will ya!

I’ve really only done minimal “tire-kicking” but I expect this could have many telcos shaking in their boots.  Not only is it super feature rich, sociable and ridiculously easy to use – it is backed by a company who has provided extremely reliable services since day one.

It is interesting to see the callout function implementation.  There have been plenty of callout scripts for Asterisk (which pretty much work exactly the same – Asteristickies is one of ‘em) but once again Google has made it that much friendlier.  Callout is a dialing function when you input a number to call, it connects to you first – then dials the remote party.

SMS is a joy! I was lucky enough to have my sister visiting this week with her unlimited text messaging plan.  I popped open the SMS box and we were sending messages with ease instantaneously.  I especially love the contact manager.  Being able to maintain a list of contacts with click to call functionality is a very powerful feature.

Anyway, I am very excited to play with my new toy – feel free to give me a ring at

(361)-GEEK-HUT

Heh… or use this thing

Follow these easy steps to point a free Washington area code phone number to your Gizmo account.  This is a great solution for those overseas who wish to have a North American DID (for free / cheap calls.)

I wrote about this app in the past, you can view the video demo here. Now I have decided to share it with others -you can find the source here.

Here is how to install it (presuming you are using Trixbox):

1. Copy the source to /var/www/html – untar it, the folder should be named services
2. Create a database mysqladmin -u root -p create services
3. cd services/
4. mysql -u root -p services < services.sql
5. nano system/application/config/database.php (match your Trixbox MySQL settings)
6. nano system/application/config/config.php and change the base_url to your Trixbox IP address / services.
7. chmod 777 -R /var/www/html/services/cisco
8. chmod 777 -R /var/www/html/services/grandstream

That should do it, you can now visit http://trixbox-ip/services and begin adding contacts / RSS feed.

Links are provided at the bottom of the application to copy and paste into your phone configurations.

This app supports GXP2000 & Cisco 79XX IP Phones.