FreePBX 2.9 Tutorial – Endpoint Manager

Foreword

I spent a couple of hours trying to configure Yealink phones via the End Point Manager in FreePBX 2.9.

Finally I get it working well and thought I should share this experience (nightmare ☺ ) with you guys.

I’m using Yealink phones in this tutorial but other brands should work just as well.

This is not a step-by-step guide but you should get enough information to setup a similar system after you went trough the steps.

1. Requirements (or what I used for this tutorial)

Debian 5.x Operating System (netinstall)
Asterisk 1.6.21.4
FreePBX 2.9.0.7

TFTP:
Make sure you installed tftpd.

On Debian you do : apt-get install tftpd

Then you create a new directory and set the rights:

mkdir /tftpboot
chmod 777 /tftpboot

After this you need to change your default tftp directory to /tftpboot by editing the file: /etc/inited.conf

And restart you tftpd server with:

cd /etc/init.d
./openbsd-inetd restart

If you have a DHCP server accessible you can configure option 66 in the DHCP server (basically add the TFTP server IP address to the DHCP settings).
If you don’t have a DHCP server you can still continue and add the TFTP server IP manually at a later state directly in the phone settings.


2. Install the End-Point Manager Module in FreePBX

After installing a basic FreePBX make sure you install the Endpoint Manager in the FreePBX Module Admin section.

This module will install a new sub-menu in under the Tools-Tab in FreePBX.

3. Configure the Main-Settings

First you need to configure some settings in the “Endpoint Advanced Settings” menu.

The section IP address of phone server refers actually to your TFTP. FTP or HTTP server where the Endpoint manager will store its files and phones connect to download their provisioning information.
In my case it’s the IP address of the Asterisk server.

Then you need to choose the protocol you like to use for your provisioning, since I installed tftpd on my Debian I choose TFTP/FTP.

After this you need to set the “Default Final Configuration Directory” in my case I set it to /fttpboot/.

This path depends on your TFTP setup; therefore you need to change the path accordingly.

4. Add phone brands and models to the Endpoint Manager

Next you go into the Endpoint Configuration Menu and Install your respective phone types. In my case all the Yealinks.

To do this select “Check for updates”. You should get a list of all supported phones.

Click the green “Install” button(s) for each brand of phone you like to install.
For each of the installations you will see a orange box displaying the installation activities. At the end of each installation you need to hit “Return”

After the installation you need to enable the phone models which you like to administrate with the Endpoint-Manager by enabling each of them.

In case you have no Internet connection to download the various phone brand XML files as described above you can download the files also from:

http://provisioner.net/releases

and then upload them via “Endpoint Advanced Settings” and then click on “Manual Endpoint Modules Upload/Export”. You will see following screen:

Here you can upload the Provisoner as well as the brand package, which you download from the above website (http://provisioner.net/releases)

5. Add a first Device

Make sure you created a working FreePBX extension before you do the following steps.

Now it’s time configure our first device (in this tutorial I use a Yealink T38P).

To do so select “Endpoint Device List” from the FreePBX menu. You have two main options to add a new device:

- Manually by entering the MAC-Address of the device
- Automatically by scanning the your network for all un-managed phones

In this tutorial I scan our network by clicking the “Go” button.
(make sure you have nmap installed on your Linux server).
The scan can take a couple of minutes, therefore just hang in there.

After a while you should see a list similar to this:

I choose the phone with the IP 192.168.2.174 and select the model T38 and assign extension 1000.

Attention: the drop down list will only show the installed and enabled phone models. So if you need other phones just go back to the step where we downloaded and enabled various phone brands and models.

After clicking “Add selected phone(s) you will see a screen like this:

Now you need to create/build the new configuration files (basically creating the files in your TFTP directory) by clicking “Rebuild Configuration for selected phones”).

This will create / update the files in your TFTP directory.

6. Configure your phone

Since I don’t have a DHCP server at my hand to configure the option 66 (basically add the TFTP server location into the DHCP communication) I need to add my TFTP server IP-Address manually into the Yealink phone.

On a T38 click on the “Phone” tab followed by “Auto Provisioning”.

In the Server URL you need to key in the TFTP server IP address.
Attention: the tftpboot directory is the default directory. Therefore you only need to key in the protocol and the IP address of your tftp server.
In my case: tftp://192.168.2.175

Make sure that “Check new config” and “Repeatly” are “On”.
Choose the frequency of the check in minutes.
Attention: the phone will only re-configure itself if there is a change in the downloaded configuration files.

Click the “Autoprovision Now” button and your phone will get the configuration files immediately or just wait until the phone connects to your TFTP server as per your Interval settings.

7. Enabling ARI module for Endpoint Manager

You can allow your users to manage certain aspects of the phone settings via the FreePBX ARI module.
The end-user can change his speed-dials, BLF, etc directly on the ARI (Recordings-Panel) in FreePBX.

This feature allows you to protect the IP phones WEB-GUI with a strong not known password.

In the “End Point Advanced Settings” you select “Enable FreePBX ARI Module” and click “Update Globals”

Then you can go to your respective Template and select which features you like to enable for self-service configuration in the ARI (Recordings Tab).

The user can now login into the Recordings module. There will be a new Menu-Point “ Phone-settings”

Attention there is a bug (spelling error in one file), which you need to change.
In most cases you will only see a blank screen after you select “Phone Settings”

Edit this file and make sure to set the paths correctly:

/var/www/html/admin/modules/endpointman/ari/modules/phonesettings.module

function action($args) {
global $endpoint, $global_cfg;

$doc_root = $_SERVER["DOCUMENT_ROOT"] .”html/admin/modules/endpointman/”;
require($doc_root . “includes/functions.inc”);

and

function display($args) {
if (!isset($_SESSION['dbh_asterisk'])) {
die(‘No Database?’);
}

$doc_root = $_SERVER["DOCUMENT_ROOT"] .”html/admin/modules/endpointman/”;
require($doc_root . “includes/functions.inc”);

You can download the PDF-Tutorial from here: Endpoint-Manager with FreePBX and Yealink.pdf

Cheers
Daniel

Asterisk 10 to come

The Evolution of Asterisk (or: How We Arrived at Asterisk 10)

Kevin P. Fleming July 21st, 2011

We are fast approaching the seven-year anniversary of the release of Asterisk 1.0.0, which occurred at the first AstriCon in September, 2004. If you look back a little further, there were various “0.x” releases made as early as December of 1999… my, how time has flown!

We’ve had quite a few ‘major’ releases of Asterisk since then, including 1.2, 1.4, and most recently, 1.8. Each of these releases has included significant changes, and sometimes architecture-improving changes. Each of them has also included substantial new functionality for Asterisk users. Along the way, we’ve been asked by many people in the community when we are going to start working on (or release) “Asterisk 2.0.” Typically, we’ve responded by saying that will not happen until we can really justify such a significant change in the version number. Many open source projects have gone through similar progressions, and quite a number of them have in fact undergone complete (or nearly complete) rewrites resulting in new ‘major’ versions.

The Asterisk project, however, has tried to avoid that level of disruption to its users. Instead we’ve focused on attempting to provide as much backwards compatibility between major releases as we could. As a result, each time we’ve released a new, major version, the decision has been made that “No, this isn’t Asterisk 2.0,” and we’ve continued with the version numbering scheme that Mark Spencer started all those years ago.

Over the past few months though, as we’ve approached the first beta release of the nextmajor version of Asterisk, we’ve been having a somewhat unexpected conversation: about just how different this release is going to be from the releases that most users in the community are using on their production Asterisk systems (primarily Asterisk 1.4, although there are still a lot of 1.2 users as well).

In fact, even though it’s been an evolutionary process, not a revolutionary one, the next major Asterisk release really will be substantially different from Asterisk 1.4 in some very noticeable ways: wideband conferencing support, basic video conferencing support, support for a number of additional VoIP technologies, full-fledged FAX support, and many others.

That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this is not Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it ‘Asterisk.’  That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, why continue to call these releases “1.x”? What does the “1″ mean, if it’s never going to change?

The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a “1.” prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the “1.” completely. The next major release, which was going to be Asterisk 1.10, will now be just “Asterisk 10″ and subsequent major releases will be “Asterisk 11″, “Asterisk 12″, and so forth.

We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon.

As always, thanks to everyone for their continued support of Asterisk. That especially includes the developer community, the people that find and report issues, the people that help test patches and the people that devote their time to answering questions on IRC channels, the mailing lists and the forums.  We hope to see everyone trying out the forthcoming beta, and we look forward to seeing you all at AstriCon 2011!

(credits: Digium Blog)

Hayibo! The first Business softphone for Asterisk (now have a Free version)

Hi All,

Intuittech Sdn Bhd, Malaysia’s largest Asterisk Installer has released Hayibo! softphone. A softphone specialized to work with Asterisk features

we wrote about this a while back but now Intuittech has released a free version of the phone ;)

some of the features include:

  • Call display and Message Waiting Indicator (MWI)
  • 2 Lines
  • 3 Way Call Conference
  • Speakerphone and Mute.
  • Redial
  • Hold
  • Do Not Disturb
  • Call history – list of received, missed, and dialed calls.
  • Call forward, Call transfer
  • Call recording
  • Video Recording
  • Managed contact list – importing and exporting contacts.
  • Acoustic echo cancellation, automatic gain control, voice activity detection.
  • Support for the following audio codecs: G.711aLaw, G.711uLaw, G.729, GSM, iLBC, G.722.1, G.722, AMR-WB, SPEEX, SPEEX-WB.
  • Support for the following video codecs: H.263, H.263+ 1998, H.264.
  • SIP compliance.
  • TLS and SRTP
  • STUN NAT traversal.
  • Support for DTMF (RFC 2833 and SIP INFO messages).
  • Auto update

as well they have added some cool Asterisk and other productivity features.

  • CRM Integration
  • SMS / Email sending from the softphone
  • Queues & Pause Codes
  • AMI Integration
  • a very nifty Voice Memo tool (record & send via email)
  • Password protected settings
  • On / Off features
  • and much more….

all in all a very complete cool softphone to be used in enterprise deployments as well call centres

you can find more information here: www.hayibo.my

cheers

Marco

Unified Communications?– Use Google and Asterisk!

Hi All,

Sanjay the CTO from Intuittech Sdn Bhd

shared some cool insights about Asterisk combined with Google voice:

1) Google Talk/Voice (and oh, jabber)

2) Calendaring

3) Festival Text To Speech Engine

we tested it and can call any google account from our deskphone for free…  isnt that great

Thank you Sanjay for sharing this.. Read the Full Article on Sanjay’s Blog

we will soon add some pictures…

Hayibo! Enterprise Business Softphone – The first Business Softphone for Asterisk

If you are in need of a great software phone look no further.

Intuittech Sdn Bhd, the winner of Digium Innovation Big Biz award 2009 brings you all Asterisk Experiences in this softphone.

check it out: www.hayibo.my

Hayibo! Standard is able to connect via AMI (Asterisk Management Interface) to Asterisk.

Currently hayibo supports:

  • Login / Logoff scenario in Dynamic agent and user setup
  • Queue login / logoff
  • Pause / Aux codes

You can setup your connection in the configuration section of hayibo under the ‘Various’ tab.
There is a button for you to test your AMI connection.

LOGIN / LOGOFF

Hayibo! fully supports dynamic agents and users login and logoff.

In a dynamic device and user setup Hayibo! will display an Asterisk login screen during the startup process, if the device is set to ad-hoc.

The user needs to login into asterisk with his username and password. This feature is comfortable and simple to use and the user does not need to remember cryptic features codes to login or logoff the Asterisk system. If the user exits the Hayibo! phone, the user will be logged off automatically from the system.

QUEUE LOGIN / LOGOFF

Hayibo! allows users respectively call center agents to easily login and logoff into Asterisk queues. The queues need to be firstly configured in Asterisk.

In the configuration settings of Hayibo! you can then configure up to 10 different queues which match your previously created queues.

The Hayibo! phone will then display a side-bar with the available queues. The agent can conveniently login to the queue by pressing one or multiple queue buttons.

As soon the agent is logged in into a queue the status will turn to red. Therefore the agents visually see in which queues they are logged in.

After an agent has logged into te queue, the pause / aux code feature is available. The agent can easily select between 10 different pause codes, which you can individually configure in the hayibo settings.

The pause codes are stored in the standard asterisk queue log and tools like asternic stats can easily display and create agent reports.

Happy Holidays

Hi All,

we are sorry, it has been really quiet on MAUG, we just had too much to do and too litlle time.

we are still looking for contributors for MAUG so if your interested please let us know. we try to update and blog more frequently the coming year that will be a really exciting one for Asterisk

so enjoy your holidays and see you in 2010

the MAUG Team

2nd Online – Workshop: How to build an Asterisk Cluster

our Sponsors from Intuit Innovation Sdn Bhd are conducting a first online-workshop via WebEx about building a Asterisk-Cluster.

This first workshop focuses on a Active-Passive setup.
(The next workshop will cover a large Active-Active setup with multiple Asterisk, DB and Load Balance servers.)

Details:

Date/Time: Thursday, 04. June 2009 at 8 AM, GMT +8 (Kuala Lumpur Time… )

Duration: 1.5 hours (depends on how many questions we have..)

max 25 People

Topics covered:

- How to choose your hardware (Servers and PSTN equipment, we will use a FoneBridge from Redfone)
- How to prepare your OS for cluster (we use Debian 5 in the workshop)
- Asterisk standard vanilla installation (we will use our install script)
- Explain the concept of DRBD and Heartbeat
- Configure DRBD and Heartbeat
- Test the cluster
- Discussion

Cost: FOC (you can buy me a beer if you are in Kuala Lumpur)

Please e-mail Daniel or Marco if you want to join the seminar and they will send you the meeting-invite!

Marco: marco (at) intuitinnovations (dot) com
Daniel: daniel (at) intuitinnovations (dot) com

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