The first step is to go into the /etc/kamailio.cfg file. In this file is the configuration for kamailio. When calls are between Kamailio extensions the call stays within Kamailio. If the call is to the PSTN then Kamailio sends it to asterisk. In the kamailio.cfg file we need to set a couple of parameters for your system. I have been building my test machine on the IP address 192.168.1.150. You will find 2 places where I have set this IP in the kamailio.cfg. Change these entries to the IP address of your machine using nano.

nano /etc/kamailio.cfg

Then restart Kamailio

service kamailio restart

Make sure kamailio is running

service kamailio status

The next step you need to do is go into freePBX. Under http://yourelastixip/admin is the freePBX interface. Go into module admin and you will find a new module called Kamailio integration. The Kamailio integration basically allows you to register handsets to Kamailio on port 5060. It also sends all external calls through asterisk when you make PSTN calls or voicemail calls. Once you have activated it go into the extensions tab in freePBX.

When adding a Kamailio extension you need to add a custom extension. If you add a SIP extension under freePBX this will simply be an asterisk extension. First add your extension number and display name. The last field required is the Kamailio password. You can scroll down and you will find an extra section called Kamailio services. Make sure it is enabled for the extension and you put in a password, click submit in the freePBX interface and reload freePBX.

The final step is to add a trunk in freePBX. Create a SIP trunk with the trunk name kamailio.

Under the peer details put

port=5060

allow=all

type=friend

context=from-internal

host=”yourelastixip”

canreinvite=nonat

This will allow calls to flow between kamailio and asterisk.

When you are done you should be able to register a phone to Kamailio with the details you put into freepbx.

Troubleshooting

If some of the packages dont install correctly try cleaning up yum

yum clean all

Then run the script again.

If everything installed correctly and you still cant make calls from Kamailio to asterisk try copying the kamailio.cfg from /var/kamailio again to /etc/kamailio. Then edit the IP address that I detailed above. Once this is done restart kamailio and try calling.

Related posts:

1. Elastix and Kamailio Part 1 – Installing
2. Elastix in the Enterprise
3. Installing FreePBXv3 in Elastix-2.0 RC3

The Kamailio module for Elastix allows you to easily integrate Elastix with a enterprise SIP server. All SIP handsets register to Kamailio. All internal SIP calls stay within Kamailio however if a call is external then Kamailio will push the call out to asterisk to deal with it.

To install the Kamalio module you first need to create the directory for it to go in. It installs itself from the directory /var/kamailio

mkdir /var/kamailio

Now change to the directory

cd /var/kamailio

Download the kamailio integration file from the MBIT website

wget http://www.mbit.com.au/kamailio-v2.tgz

Untar the file into the directory

tar zxvf kamailio-integration.tgz

In the directory now will be a heap of files you want to execute the script that installs all the modules for you. To run the script execute install_kamailio.sh

./install_kamailio.sh

Soon as it is installed we will move to the next part for configuring Kamailio. The installation can take a while so please be patient. The installation can only be used with Elastix 2.0.

Related posts:

1. Elastix and Kamailio Part 2 – Integrating
2. Elastix in the Enterprise
3. Installing FreePBXv3 in Elastix-2.0 RC3

Related posts:

1. Elastix: 1.2 Stable Released
2. Asterisk 1.8.0-Beta2 Now Available
3. Asterisk 1.6.1.13 Now Available

This is still very much a work-in-progress, as we’re merging two unfinished products and I’ve not had enough of a chance to test it thoroughly either so I’m hoping for some response from the community on this one. I think this is quite possibly the first time that FreePBXv3 has been tied into a major distro, and I reckon it’d be cool to see Elastix-2.0 support FreePBXv3 officially.

I have however tested Ext’s registering happily, calling each other, so I know that FreePBXv3 is tied in correctly.

Anyway, I’ve tar’d up a slightly older version of FreePBXv3, as they’re undergoing some design changes right at the moment which mean that the current version doesn’t install. So, we’ll be using this.

Total installation should only take about 5 minutes, so this How-To will be nice and short!

So what you’re going to do is start by installing the latest Elastix-2.0 RC3. In my case I used i386.

Once it’s installed, ssh into the server and run:

cd /var/www/html
wget http://c2s.co.nz/dumpfiles/freepbxv3.tar.bz2
tar xvjf freepbxv3.tar.bz2
cd freepbxv3
sh preinstall.sh

It’s 8.5MB, and this is the snapshot of FreePBXv3 that we’re going to be using.

I’ve already modified the preinstall.sh so it uses the user “asterisk” instead of “nobody”, keeping in line with how Elastix does things.

You’re going to then run the following commands to upgrade PHP:

yum install yum-priorities
cd /etc/yum.repos.d/
wget http://dev.centos.org/centos/5/CentOS-Testing.repo
yum --enablerepo=c5-testing install php
/etc/init.d/httpd restart

We need PHP-5.2 because of one of the FreePBX dependencies (Kohana) and Elastix only comes with PHP-5.1

Now you need to fire up the Elastix WebUI, but instead of just going to the regular URL, add in /freepbxv3/ to the end and you should be greeted with the install window:

Agree, hit Next.

You will be prompted with a pre-flight checklist. It should pass the lot. Again, hit Next.

Now, on the next screen, you’re going to see the database details.

Set the database to freepbxv3, the username to “root” and put in the default Elastix root MySQL password: eLaStIx.2oo7

You might also want to adjust your timezone.

Now, create a Username and a Password that you will use to access the system. Whatever you enter in here will be the email address used to log you in, it’ll create the account based off this, so remember the address / password you use. Don’t worry, this account is unique to your Elastix / FreePBXv3 installation.

On the next screen you need to change it from the FreeSWITCH driver to the Asterisk driver. All the other defaults are fine except the AMI admin password which needs to be changed to “elastix456″

The next screen will complain a little about a few bits n pieces relating to Media Management. We’re going to ignore this for now simply because I’m lazy. Scroll right the way down and hit “Next”, as the default selections suit us fine.

It will then go through and do the installation of everything:

When that’s done you’ll be taken to your newly installed FreePBXv3 system!

One thing I’ve found is that the default passwords didn’t seem to work until I modified the password on the accounts. You can do this under Destinations –> Devices

Select Device 1 and choose Edit. Adjust the password and then hit Save.

You should now be able to take a phone and register it using the Username “sip1″.

I then did the same for Device 2, registered with “sip2″ after changing the password, and was able to make calls freely between them.

Granted there’s a lot more to be tested, but that should be enough to get you pointed in the right direction. I’d love to hear from some people how they get on with this, and the blog will be updated in-future.

Related posts:

1. Elastix and Kamailio Part 1 – Installing
2. Installing ADA (Asterisk Desktop Assistant) on Elastix
3. First Screenshots of Elastix 2.0

I am looking to develop a new endpoint manager. It will look to include the following features

• Provision many different vendors such as Linksys/Cisco, Snom, Aastra, Yealink, Polycom
• Provision all functions on each phone. This includes extension, all BLF buttons and regional settings
• All settings held on the Elastix machine in a mysql database
• Phones will grab all their settings TFTP
• Endpoint manager will be able to scan for extensions and provision
• Endpoint manager will be able to create a TFTP file so that the phone can be sent to site and it will be completely provisioned remotely
• Multilanguage support

What does this mean for you?

Vendors

With a good endpoint manager this will mean less support issues for vendors. It will make it easier for users to setup and use their IP phones. Vendors will have to do less support and the users of their phones will not have to have high end technical skills to use the best features of the phone.

Elastix Users

This will save you time, whether you are installing 10 phones or 100 phones. It will also make setting up phones very easy.

What is needed from Vendors and Users?

We need you to contribute. To build a decent module that will support many different types of phones we are looking at a budget of about $10000.00. If everyone can get together we will be able to build this and it will make installing Elastix better for everyone. All code developed for this project will be given back open source to the Elastix project.

Contributors

I have put together a scale of contributing. Not everyone can contribute a lot of money but if everyone can put in then it will help us get to the goal of $10000.00. The following sponsorships are available

Gold Sponsorship – $1000.00

Silver Sponsorship – $500.00

Bronze Sponsorship – $100.00

I am behind this project 100% and have put up $1000.00 to get the ball rolling. To donate to the project you can contribute to the paypal account – elastixprojects@mbit.com.au. Once you have contributed please let me know via email on mark@mbit.com.au

Total remaining – $8900.00

Contributors

MBIT Technologues – $1000.00

Sunshine Networks – $100.00

Thank you for all who contribute. We are looking to make a great new feature for Elastix that will benefit everyone.

Related posts:

1. Elastix in the Enterprise
2. New Elastix: version 1.1 available
3. Elastix: 1.2 Stable Released

For this I wanted to be able to have Kamailio as my SIP server but have asterisk provide all the features it has such as voicemail. For this a FreePBX module was created that brings together Kamailio and asterisk. How it works is everything registers to Kamailio. All internal calls stay within Kamailio and if any calls created are not internal then it sends it to asterisk to manipulate. All the extensions are created in FreePBX which is handy because this means you dont have to keep a record of the extension in asterisk and Kamailio.

This allows you to take your Elastix machine from handling around 200 extensions to up around 1000 with no sweat.

Related posts:

1. Elastix and Kamailio Part 1 – Installing
2. Elastix and Kamailio Part 2 – Integrating
3. Elastix: 1.2 Stable Released

That’s great if you have ABE, but not so good if you’re one of the millions of installs out there that uses vanilla Asterisk.

This guide presumes you already have ADA downloaded and installed. If not, you can get the latest (At the time of writing) version 1.1 from here:

http://dl1.digium.com/ADA1.1/ADAInstall.exe

Now you’re going to ignore any instructions you’re previously read, and we’ll start from scratch. Don’t worry, you’ll be up and running in a matter of moments!

Once ADA is downloaded, installed and running, you should be greeted with a login window.

What you now need is the “manager” user to be setup.

The “manager” user is different from your regular extension, in that it allows you to control the actions of Asterisk. With this in mind, you need to be careful when setting up an Ext, not to have a weak password, and make sure you use the permit / deny lines, even for remote extensions. You can get a password generated for you here: http://www.thebitmill.com/tools/password.html

From the shell, we’re going to edit /etc/asterisk/manager_custom.conf like this:

nano -w /etc/asterisk/manager_custom.conf

You’re now going to create a new entry with the following details, in my case I’m Ext 199:

[199]
secret=mkHHa031kHMsGRu
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

Now you need to create a manager user for every Ext that will be using ADA. This is because ADA uses the username part as the Ext for it to ring to connect the call to.

Without it, it would be the same as you dialing the number, and the system just sitting there, holding it.

Ctrl + X to quit nano, and you can safely get your ADA desktop client to login with the username: Ext@asteriskIP

In my case it would be something like: 199@192.168.0.250

Use the password you specified above in the manager user, and not your regular SIP Secret.

If you find you’re unable to connect, confirm that you’ve got the right password, and adjust the “permit=” line as applicable. I would highly recommend leaving it in there and just adjusting it for your IP range. Same goes for external or remote clients, make sure that you have their IP Address.

Now that you’ve done that, you need to hook ADA into your dial-plan.

Again, the standard documentation is no good here, so, we’re going to edit another file:

nano -w /etc/asterisk/extensions_custom.conf

At the top of this file (Well under the first tow lines of comments) we’re going to add in the following:

[ada]
include => from-internal

Then hit Ctrl + X and follow the prompts to save and quit.

Again the standard documentation is no good for us there, as the “ADA” context is actually “ada” in lower case, and we’ll use the “from-internal” instead of “default”.

Now you need to refresh the dial-plan (Just this once) so run this:

asterisk -rx 'dialplan reload'

You’re now all good to go! On your ADA desktop client, you can create a test call! It’s easiest to test to another Ext in my opinion, so enter in the Ext # of another extension on the system and hit “enter”.

What should happen:

Your phone will ring immediately

You pick it up and will hear ringing, the destination phone should now also be ringing

Easy!

If this has helped you in any way, please leave a comment and say Hi.

Cheers

Chill.

Related posts:

1. Installing FreePBXv3 in Elastix-2.0 RC3
2. Elastix and Kamailio Part 2 – Integrating
3. Using the Patton SN4554 for ISDN with Elastix

This basic How-To shows you how to set it up to work with your Elastix system:

First off, you’ll want the following config which is a bit of a mash-up from some other documentation on making it work with an Aastra 800 (Include the header):

#----------------------------------------------------------------#
#                                                                #
# Template for SN4554/2BIS/UI to use with Elastix                #
# Use with firmware R5.1 or higher                               #
#                                                                #
# This template uses dhcp to retrieve an IP address. Comments in #
# the file indicate what to change (Start with '#')              #
#                                                                #
# See the SmartWare Software Configuration guide for command     #
# details (http://www.patton.com/manuals/SCG-r52.pdf)            #
# Modified by Josiah Spackman                                    #
# http://chillingsilence.wordpress.com                           #
#----------------------------------------------------------------#

cli version 3.20
webserver port 80 language en

system

  ic voice 0
    low-bitrate-codec g711alaw64k

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile ppp default

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  fax transmission 1 relay t38-udp

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
# replace 'dhcp' with your fix IP if needed, e.g. "ipaddress 172.16.1.20 255.255.0.0"
    ipaddress dhcp
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
# uncomment the following line if you need to add routing table entries
  # route 0.0.0.0 0.0.0.0 172.16.1.1

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_ISDN_TO_SIP
# 'T' in the following lines specifies "digit collection". The time for collection can be adjusted replacing 'T' wih 'T2' for 2 seconds.
    route T dest-interface IF_SIP

  routing-table calling-e164 RT_SIP_TO_ISDN
    route default dest-service SV_HUNT_PSTN MP_Unknown-Subscriber

# This mapping table sets the ISDN type of number for calls towards ISDN to 'subscriber'
  mapping-table calling-e164 to calling-type-of-number MP_Unknown-Subscriber
    map default to subscriber

  interface isdn IF_ISDN_0
    route call dest-table RT_ISDN_TO_SIP

  interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP

  interface sip IF_SIP
    bind context sip-gateway GW_SIP
    route call dest-table RT_SIP_TO_ISDN
# This is the IP of your Asterisk. Replace with domain name if DNS server is available
    remote 192.168.0.250

  service hunt-group SV_HUNT_PSTN
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_ISDN_0
    route call 2 dest-interface IF_ISDN_1

context cs switch
  no shutdown

# The parameters realm, username and password have to match your Asterisk configuration
authentication-service AUTH_AASTRA800
  realm 1 smartnode-gw
  username patton password 6953789

# The parameters domain, username, identity (=username) below have to match your Aastra800 configuration
location-service LS_AASTRA800
  domain 1 smartnode-gw

  identity-group default

    authentication inbound
      authenticate 1 authentication-service AUTH_AASTRA800 patton 6953789

  identity 6953789

    authentication inbound
      authenticate 1 authentication-service AUTH_AASTRA800 patton 6953789

    registration inbound

context sip-gateway GW_SIP

  interface WAN
    bind interface IF_IP_WAN context router port 5060

context sip-gateway GW_SIP
  bind location-service LS_AASTRA800
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface IF_IP_WAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_ISDN_0 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_ISDN_1 switch

port bri 0 1
  no shutdown

Copy this all into a notepad window and save it as “SN4554.cfg”

What you’re going to want to modify is the references to “192.168.0.250″ and point it to your Elastix / Asterisk system.

All the ISDN ports are set to “Point-to-Point”. You can adjust the “pp” settings to “pmp” if you require “Point-to-Multipoint”, though to be honest ISDN isn’t my strong point so I’m not entirely sure how relevant that is.

This will leave the WAN port using DHCP, hopefully you’ve got a semi-intelligent DHCP server that will give out the same IP each time.

Open the WebGUI of your SN4554 up, login with the default Username “administrator” and a blank password.

Click on “Import / Export” on the left-hand side, then hit the Import Configuration tab.

Select the file, import it, then reload the device.

Now, in asterisk you want to add a new SIP trunk. Name it “ISDN” and put the following in the SIP PEER Details:

username=patton
type=friend
secret=6953789
qualify=1200
insecure=very
host=192.168.0.141
dtmfmode=RFC2833
disallow=all
context=from-pstn
canreinvite=no
allow=alaw&g729

There’s no register string or USER details, though it may be worth specifying the max channels as “4″.

You’ll need to adjust the “host” setting to the WAN IP Address of your Patton box, as we’re semi-insecure with such a basic password.

You *could* adjust the password and username in the config etc as applicable, but specifying the host should be enough, especially considering (in my instance) the whole system is LAN-accessible only.

When you’re finished, you should be able to login to your Elastix system via SSH and run:

asterisk -rx ’sip show peers’

And see:

ISDN/patton                192.168.0.141        N      5060     OK (19 ms)

If you do, congratulations, you’re ready to go!

Now you just need to setup inbound and outbound routes as applicable.

If this was useful to you, please leave a comment and say hi.

I’d also like to thank Byron from SnapperNet in New Zealand, he’s been such a great help, and also provided me with the initial configuration samples.

This How-To has also been re-posted here: http://chillingsilence.wordpress.com/2010/05/06/using-the-patton-sn4554-for-isdn-with-elastix/

Related posts:

1. Elastix and Kamailio Part 2 – Integrating
2. Installing FreePBXv3 in Elastix-2.0 RC3
3. Installing ADA (Asterisk Desktop Assistant) on Elastix

Related posts:

1. Elastix: 1.2 Stable Released
2. Elastix 0.7 is here!
3. Elastix 1.4 released

Elastix - The reliable PBX appliance software

Dear Elastix community,
We are proud to announce the release of Elastix 1.6 stable. It can be downloaded in 32-bit or 64-bit ISOs. This release has several improvements from Elastix 1.5, that would fix bugs from Elastix and CentOS. Here are some of the more important changes:

• Asterisk updated to 1.4.26.1-4
• Dahdi updated to 2.2.0.2-6
• Asterisk now supports Huawei modems as trunks. Thanks to Odicha , creator of chan_sebi
• Kernel updated to 2.6.18.164.el5, the latest CentOS 5.3 kernel

Read the complete changelog

Related posts:

1. Elastix 1.4 released
2. First Screenshots of Elastix 2.0
3. Elastix 0.7 is here!

Elastix - The reliable PBX appliance software

It has been a while since the release of Elastix 1.0. There have been a lot of ideas since then that were scheduled for the next release. Meanwhile, we released the Call Center and Developer modules as GPL and a lot of other useful add-ons. These releases have brought satisfaction to all of us in Elastix. Our community has grown and our downloads are almost half a million in total.

Finally, the wait for Elastix 2.0 is over, and time for new ideas has come. We have started development of this new version and we will try to include as many ideas and suggestions from the community as possible.

For those of you that just can’t wait, here is a preview of some of the main features that have been scheduled for Elastix 2.0:

• Asterisk 1.6
• Elastix Operator Panel release
• Meeting / Presentation sharing
• Advanced reports
• Improvements to the Call Center Module
• Better integration between all components

Asterisk 1.6

This is a very important migration that we’ll be working on. Our focus will be mainly on taking advantage of Asterisk 1.6′s performance improvements.

Elastix Operator Panel

The same old Flash Operator Panel (FOP) will still be available. Nevertheless we have decided to develop our own alternative, specially now that FOP version 2 would not be open source and that is not compatible with the GPL license that Elastix uses.

It is our hope that this alternative Elastix Operator Panel will be available for any distro in the future including those not based on Elastix.

The Elastix Operator Panel will be Ajax-based instead of Flash-based. Our main objective will be performance improvement.

A New Way to do Meetings

Since the release of Elastix 1.0, we have seen how meeting/conference solutions have improved. A clear example of this is the success of Webex^tm. Elastix will stay follow the latest trends that the latest communications technology demands.

The development of this feature will be divided in two phases. The first phase will be included in the Elastix 2.0 release and will integrate telephony, chat, file sharing and document presentations. The second phase will add further functionality gradually, after the release of Elastix 2.0.

We are sure that the implementation of this feature will maintain Elastix as the leader among Unified Communications distros worldwide.

Advanced Reports

One of the strongest features of Elastix is our reports. Our development framework has improved and will now allow us to improve our reports even more.

In addition to improving data reports, they will be nicer looking and friendly. The new Elastix framework, based on PHP, includes Ajax-management libraries that will allow us to improve user interaction, among other things.

Improvements in the Call Center Module

Elastix has been a pioneer in releasing a fully-functional open source call center solution. Other distros have chosen to close and market their solutions. However, Elastix gives you FREE download of this module under GPL.

The Call Center module gives Elastix insights in the area of large installations.

Among the improvements scheduled for this module, we will work on streamlining the installation and update processes, and performance upgrades. There will be better integration between the communications components of Elastix 2.0 such as PBX, IM, Fax, Email, Agenda and others. We will also focus particularly on the integration of the Agenda with the rest of the components. This will help boost efficiency in the companies that use Elastix.

Of course this is just a sample of the new features in Elastix 2.0. We hope that Elastix 2.0 will see more contributions from the community so we can include as many of your suggestions as possible.