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.

I have been busy integrating Asterisk into NimbleBilling – take a look see and let me know what you think!

I stumbled upon this module last night when trying to delimit access to an outbound route per extension. I added a colleague to my PBX who had purchased an account at voip.ms; problem was when he would dial out it would always display my Caller ID and use my trunk (which is bad if he dials international). I guess this is a frequently asked question so I created a video demonstrating the great power of this module – you will be surprised how easy creating permissive contexts can be This video will show you how to install the module and limit access to outbound routes / trunks per extension. The goal is to partition an Asterisk IP PBX in order to host multiple tenants on one system.

The following information is from this site.  Please check it out for download link and changelog / usage.

Possible Uses

• Restrict access to certain outbound routes or feature codes by a particular extension or group of extensions.
• Give particular extension(s) priority access to certain outbound routes, such as a particular emergency route associated with their geographic location.
• Give certain outbound routes top priority for use during “free” or low cost calling periods, while making those same routes lower priority (or disallowing access entirely) during higher cost time periods.
• Disallow access to outbound routes (with possible exception of Emergency access) to certain (or all) extensions during particular time periods (don’t let night cleaning crew make long distance calls, or disallow outgoing night calls from telephones in children’s rooms, while still allowing emergency number calls).
• Allow two or more families/companies/organizations to use the same FreePBX box, while still allowing each to have access only to “their” outgoing routes and trunks.
• If you have a SIP provider that does not send DID (normally a pain to handle because you can’t create a normal Inbound Route), set up a new custom context (call it idiot-provider), give them no access to anything (deny all), and then specify where you want their calls to go in the Failover Destination. Then put context=idiot-provider in that provider’s trunk user details.

Module Description

One feature which was a bit lacking in Asterisk/FreePBX was the ability to easily create multiple tenants.

This module creates custom contexts which can be used to allow limited access to dialplan applications.

Now allows for time restrictions on any dialplan access!

This can be very useful for multi-tenant systems.

Inbound routing can be done using DID or zap channel routing, this module allows for selective outbound routing.

House/public phones can be placed in a restricted context allowing them only internal calls.

Custom contexts can now be used as destinations. An IVR menu, Time Condition, etc. can now send a caller into a custom context. This feature requires FreePBX 2.2.0rc2 (or the latest SVN version if prior to the release of rc2)

(The following are the module author’s comments, “I” refers to the module author, not the original creator of this wiki page).

A number of improvements have been made to freePbx to handle multiple tenants.

1) inbound routing based on zap channel – i used to have to hack it by putting each zap channel in its own context.

2) authtype = database allows for dividing extension ranges

the main problem for me was outbound routing…

I wanted some extensions to dial out one route, and others out another route.

I had to create a custom context for each, then place each in their own custom context, then include all of the contexts which they should have access to. This became a nuisance as each module added its own context to from-internal-additional which could not be included as it also contains outbound-allroutes.

The purpose of this module is to dynamically list all contexts included in any contexts you choose, and allow you to create custom contexts which can include any of these all without config editing.

I JUST completed the FreePBX module for ENUMPlus and our DNS is up and kicking ass – check out the video and Sign Up!

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.