Glenn: catchg

2015-10-26T10:38:28Z

catchg

Admin at 03:41, 17 June 2015

2015-06-17T03:41:02Z

Admin: Created page with "''By AdNovea – Mars 2009'' {| border="0" |- valign="top" |
Asterisk logo '''TOC''' | | = HOWTO on Asterisk IP-PABX* (SIP/I..."

2015-06-17T03:27:19Z

Created page with "''By AdNovea – Mars 2009'' {| border="0" |- valign="top" | Asterisk logo '''__TOC__''' | | = HOWTO on Asterisk IP-PABX* (SIP/I..."

New page

''By AdNovea – Mars 2009''

{| border="0"
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| [[Image:Asterisk.png|Asterisk logo]] '''__TOC__'''
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= HOWTO on Asterisk IP-PABX* (SIP/IAX VoIP) =

*''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. 

{| cellspacing="1" cellpadding="1" border="0"
|- valign="top"
| <blockquote>
*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 
</blockquote>
|
*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

|}

|}

 

Students have a good opportunity to type [http://supremeessays.com term papers online] and get help.

= Foreword =
<blockquote>'''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'')
 [[Image:Exclamation.png]]'''IMPORTANT:'''

On your router forward the port '''5060''' to your NAS server

Asterisk Web interface default ID/PWD is '''admin/password''' </blockquote>
 

'''Source(s):''' [http://www.downloadranking.com Asterisk IP-PABX*(SIP/IAX VoIP)]

= Installation of ASTERISK =
<blockquote>
== QPKG installation ==

*Download, install and enable the Asterisk QPKG
*Click the Web page hyperlink (''http: //<your_nas>:8088/asterisk/static/config/cfgbasic.html'')
*Enter the default A/C (''admin/password'') and you will be prompted to change the password right away
*Under '''Dial plan''' create a new dial plan (''see below '''Configure your Asterisk PBX server''''')
*Under '''Users''' create two new users
*Install 2 soft phones (''see below '''Testing the PBX''' '')

 

== Manual installation on x86/ARM platforms ==
<pre># ipkg update
# ipkg install asterisk14
# ipkg install asterisk-gui
# ipkg install asterisk-sounds
</pre>
'''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: 
<pre>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
</pre> </blockquote>
 

= Configure ASTERISK (''manual installation'') =
<blockquote>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)'' <pre>enabled = yes
webenabled = yes
httptimeout = 60

[administrator]
secret = mypassword
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
</pre>
Then you have to modify '''/opt/etc/asterisk/http.conf''':
<pre>enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
</pre> </blockquote>
 

= Start & Stop Asterisk =
<blockquote>
== Using a SSH console ==

Asterisk can be launched as a deamon or with the CLI (''Command Line Interface'')

=== IPKG installation (''manual installation'') ===
<pre># /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)
</pre>
=== QPKG installation ===
<pre># /usr/sbin/asterisk -vvvgc -C /etc/asterisk/asterisk.conf
# /usr/sbin/asterisk -r -C /etc/asterisk/asterisk.conf
...
</pre>
== Using the QPKG page ==

*Enable or Disable Asterisk from the QPKG administration page
</blockquote>
 

= Asterisk web GUI administration =
<blockquote>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.html''' '''NOTE''': ''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.
</blockquote>
 

= Configure your Asterisk PBX server =
<blockquote>
== 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. [[Image:Asterisk Dialplan.jpg|thumb|center|500px]] 

== Create new users ==

In order to test our Asterisk PBX, we need at least 2 users. Create the users with the here below parameters
<pre> CallerID: 6000 and 60001
Name : User1 and User2
DiaPlan: DialPlan1
Enable Voicemail for this User: Checked
</pre>
[[Image:Asterisk Users.jpg|thumb|center|500px]] 

== Configure Voicemail ==

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'')
<pre>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
</pre>
 

 

== 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.
</blockquote>
 

= Testing the PBX =
<blockquote>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. [[Image:Asterisk 3CX-debug.jpg|thumb|center|500px|Asterisk 3CX-debug.jpg]]
</blockquote>
 

= 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
<blockquote>
== 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.

 Here is an sample of [http://forum.qnap.com/viewtopic.php?f=24&t=3592&p=62431#p62431 providers.conf] for some ISPs.

[http://www.whichvoip.com WhichVoIP] also offers free [http://www.whichvoip.com/voip/sip-trunking-providers.htm SIP Trunking Provider] comparisons and information ''Feel free to correct/update this list.'' 

== VOIP trunks ==

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.
</blockquote>
 

= Remove Asterisk (''IPKG installation'') =
<blockquote>
== After manual installation ==
<pre># /opt/sbin/asterisk -rx 'stop now'
# ipkg remove asterisk-sounds
# ipkg remove asterisk-gui
# ipkg remove asterisk14
</pre>
*Uninstall Asterisk QPKG from the QNAP Admin page (''this is a small QPKG'')

== Asterisk QPKG ==

*Uninstall Asterisk QPKG from the QNAP Admin page (''this is the "official" QPKG'')
</blockquote>
 

= More… =
<blockquote>Need to configure a softphone, an IP Phone, the voicemail, etc … http://www.asteriskguru.com/tutorials/</blockquote><blockquote> </blockquote>
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[[Category:Communications]] [[Category:IPBX|I]] [[Category:Asterisk|A]]

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 *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'')
-<br> [[Image:Exclamation.png]]<span style="color: darkred">'''IMPORTANT:'''
+<br> [[Image:Exclamation.png]]<span style="color: darkred">'''IMPORTANT:'''On your router forward the port '''5060''' to your NAS server Asterisk Web interface default ID/PWD is '''admin/password'''</span> </blockquote>
 <br>

Asterisk - Revision history

Glenn: catchg

Admin at 03:41, 17 June 2015

Admin: Created page with "''By AdNovea – Mars 2009'' {| border="0" |- valign="top" | Asterisk logo '''__TOC__''' | | = HOWTO on Asterisk IP-PABX* (SIP/I..."

Admin: Created page with "''By AdNovea – Mars 2009'' {| border="0" |- valign="top" |
Asterisk logo '''TOC''' | | = HOWTO on Asterisk IP-PABX* (SIP/I..."