Internet Protocol Private Automatic Branch eXchange aka IP-PBX or IPBX
Asterisk-based telephony is a versatile IPBX with tons of features (see below!) . It supports classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems.
ADSI On-Screen Menu System
Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number Discovery (DUNDi™)
Do Not Disturb
E911
ENUM
Fax Transmit and Receive (3rd Party OSS Package)
Flexible Extension Logic
Interactive Directory Listing
Interactive Voice Response (IVR)
Local and Remote Call Agents
Macros
Music On Hold
Music On Transfer:
- Flexible Mp3-based System - Random or Linear Play - Volume Control
Predictive Dialer
Privacy
Open Settlement Protocol (OSP)
Overhead Paging
Protocol Conversion
Remote Call Pickup
Remote Office Support
Roaming Extensions
Route by Caller ID
SMS Messaging
Spell / Say
Streaming Media Access
Supervised Transfer
Talk Detection
Text-to-Speech (via Festival)
Three-way Calling
Time and Date
Transcoding
Trunking
VoIP Gateways
Voicemail:
- Visual Indicator for Message Waiting - Stutter Dialtone for Message Waiting - Voicemail to email - Voicemail Groups - Web Voicemail Interface
Zapateller
Computer-Telephony Integration
AGI (Asterisk Gateway Interface)
Graphical Call Manager
Outbound Call Spooling
Predictive Dialer
TCP/IP Management Interface
Scalability
TDMoE (Time Division Multiplex over Ethernet)
Allows direct connection of Asterisk PBX
Zero latency
Uses commodity Ethernet hardware
Voice-over IP
Allows for integration of physically separate installations
Uses commonly deployed data connections
Allows a unified dialplan across multiple offices
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Foreword
SIP (Session Initiation Protocol) is a protocol for VoIP (Voice over IP) used to carry audio exchanges between users through the Internet infrastructure rather than hardwired telephone lines (PSTN). In order to do so there two requirements:
Need to have the relevant pieces of software to handle this protocol (it is implemented into the ISP boxes)
Need to have a capability to connect from Internet to the PSTN (handled by your ISP when you have the capability to use VoIP with your ISP boxes)
What we suggest to do here is:
To first implement on a QNAP server a local PBX (such as an old telephone switchboard) to support several phone lines. The handsets may be either softphones on PCs or IP-Phones.
To connect our internet phone network to the outside using our ISP SIP capability.
There are 2 possibilities to install Asterisk on your NAS
Through the Asterisk QPKG available from the QNAP forum
Using the IPKG Manual installation (requires IPKG-Optware QPKG to be installed and enabled on your NAS)
IMPORTANT:On your router forward the port 5060 to your NAS server Asterisk Web interface default ID/PWD is admin/password
NOTE: I recommand to select the GUI v2.0 as mentioned above. There is also a "asterisk14-gui" package for the GUI 1.4. but it may lead you to the wizard and lock. You can come back to the login page and select from the left panel the actions to perform but you may also encounter issues to create users. The asterisk16 package has some issues and the GUI is keeping reloading - There are some known bugs and until the ipkg will not download the fixed v1.6 version, we recommand to use the v1.4
IMPORTANT: with the IPKG installation, replace "/usr" by "/opt"
Save the default configuration files
It is recommended to back up your configuration files before you continue. To achieve this just copy /etc/asterisk under different name:
cp -r /opt/etc/asterisk /opt/etc/asterisk.backup
NOTE:
on X86 /opt = /share/xx0_DATA/.qpkg/Optware/
on ARM /opt = /share/xx0_DATA/optware/opt
Configure ASTERISK (manual installation)
There are mainly two files that you should modify. The /opt/etc/asterisk/manager.conf file will be modified and a new user added. The administrator user will enable the Asterisk Administation through the GUI. (much easier than editing the files for a begineer)
Then you have to modify /opt/etc/asterisk/http.conf:
enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
Start & Stop Asterisk
Using a SSH console
Asterisk can be launched as a deamon or with the CLI (Command Line Interface)
IPKG installation (manual installation)
# /opt/sbin/asterisk –vvvgc Start in debug mode with the CLI
*CLI> stop now Stop Asterisk from the CLI
*CLI> module reload Restart Asterisk (for example after a file configuration modification)
# /opt/sbin/asterisk Start as a deamon
# /opt/sbin/asterisk -rx 'stop now' Stop Asterisk (-rx sends a CLI command)
# /opt/sbin/asterisk -r Open the CLI console (type quit to exit the CLI)
Enable or Disable Asterisk from the QPKG administration page
Asterisk web GUI administration
In order to be able to load the asterisk-gui, the configuration files must be modifies as stated and the Asterisk server restarted. Then you can use asterisk-gui: http: //<your_nas>:8088/asterisk/static/config/cfgbasic.htmlNOTE: When the "Advanced options" is enable in the GUI, a CLI console will also be available from the GUI.
Backup the Asterisk configuration
In the left panel click on Backup then on Download from Unit. You can save your configuration files on your local disk for future restore if needed.
Required details: How to reload the saved file from the PC to Asterisk for restoring the settings.
Configure your Asterisk PBX server
Create profiles (DialPlan)
You may have different types of phone usages to manage (conference rooms, employees, guests, etc.). Each will have different privileges to access or not some areas such as international calls. The profiles are stored in DialPlans. To create the first profile, click on Dial Plans and Save. Leave options as is for this very first profile.
Click on Voicemail on the left panel and setup as follow: (we use 999 for the because it's the one used by default our softphones)
Extension for checking messages 999
Direct Voicemail Dial checked
Say message Caller-ID checked
Say message duration checked
Play envelope checked
Allow users to review checked
Advanced options
Set the language
On the left panel select Options and click on the tab "Languages". Select the preferred language.
Display Advanced options
On the left panel select Options and click on the tab "Advanced options". Click on "Show Advanced Options", additional actions are displayed at the bottom of the left pane.
Testing the PBX
Install on two (Windows) computers two softphones for examples. We have used 3CX softphones (http://www.3cx.com).
Configure each phone numbers extension 6000 and 6001.
Set an ID (name of PC user for example)
Click on “Advanced setting” and check “I’m in the office – local IP” and enter your nas IP address
Click on button "Appy Changes". If you forget this step you get error message: "Not connected: Destination not found"
You must be able to talk to each other and do much more like leaving message, parking a call, etc…
Debugging
Shall you encounter issues with Asterisk and 3CX, you may enable the debugging window to get some information regarding the problem.
Connect your PBX to the outside (add a Service provider trunk)
From the left panel, select Trunks and a new page is displayed with four tabs. There are 4 possibility to connect your IPBX to the outside:
using an external hardware box to connect to a standard PSTN line
using the VOIP capability and pre-setting of your ISP box if listed
setting manually the SIP/IAX parameters for your trunk using the VOIP capability of your ISP box
using an hardware to use your connection to support VOIP
Analog trunk
Depend of your external hardware. Select "Configure Hardware" from the Asterisk Control Panel left pane and setup the Analog Hardware parameters.
Service providers
The list of providers is stored under /etc/asterisk/providers.conf. By default such a list is empty. The provider parameters may be listed in the "providers.conf" file for easy future installation of the trunk. I have created (but of course, I have no way to test all the configuration inside) a list to help the trunk configuration.
Depend of your ISP. Select the type (IAX/SIP), enter Provider Name, Hostname, Username and Password. It assumes the "Provider name" parameters to be define somewhere else in the configuration files.
T1/E1/BRI Trunks
Depend of your digital hardware. Select "Configure Hardware" or "mISDN Config" from the Asterisk Control Panel left pane and setup the parameters.
IMPORTANT:On your router forward the port 5060 to your NAS server Asterisk Web interface default ID/PWD is admin/password
