Showing posts with label AsteriskNOW / FreePBX.. Show all posts
Showing posts with label AsteriskNOW / FreePBX.. Show all posts

Installing Webmin On AsteriskNOW

Posted: Thursday, June 13, 2013 by Unknown in Labels: ,
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What is Webmin? 
Webmin is a web-based interface for system administration for Unix. Using any modern web browser, you can setup user accounts, Apache, DNS, file sharing and much more. Webmin removes the need to manually edit Unix configuration files like /etc/passwd, and lets you manage a system from the console or remotely.

Installing the RPM :
1.Connect to AsteriskNOW using SSH,download the RPM package.
wget http://prdownloads.sourceforge.net/webadmin/webmin-1.630-1.noarch.rpm

2.Install the RPM package.
rpm -U webmin-1.630-1.noarch.rpm

The rest of the install will be done automatically to the directory /usr/libexec/webmin, the administration username set to root and the password to your current root password. You should now be able to login to Webmin at the URL http://localhost:10000/. Or if accessing it remotely, replace localhost with your system's IP address.

Installing The Free G729 Codec For AsteriskNow

Posted: Saturday, June 1, 2013 by Unknown in Labels: ,
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G729 is a low bandwidth codec, which can operates at a lower bandwidth of 8Kb/s per call instead of 64kb/s when using G711a-law /G711u-law. This describes how to install the G729 free codec on AsteriskNow without the need to license it from Digium Addons.


 Installing The Free G729 Codec For AsteriskNow:

1-Infomation you need to gather to download the proper codec .SO binary file.
2-Download the G729 codec source or binary.
3-Rename the binary file.
4-Give it the necessary permissions.
5-Load the module into Asterisk.
6-Verify the module has loaded correctly.
7-Configure your phones to use G729 instead of choosing the G711a-law /G711u-law.
8-Verifiy the codec usage by your phones on your asterisk server.

1-Infomation you need to gather to download the proper codec .SO binary file.
-Processor Architecture: Check the kernel type installed on your system.It's 64 Bit OS.

-Asterisk Version: As you can see here the Asterisk version is V11.2.1.


2-Download the G729 codec source or binary.
-To download the codec binary click here.and choose the codec source / binary  that is compatible with your Asterisk version and architecture.
-Go to the modules directory in " /usr/lib64/asterisk/modules " .


-Download the codec file using the " wget " command in the " /usr/lib64/asterisk/modules " directory.



3-Rename the binary file.
-Now copy or move the binary to rename it using the "cp codec_g729-ast110-gcc4-glibc-x86_64-core2.so  codec_g729.so "

4-Give it the necessary permissions.
-Use the " chmod +x codec_g729.so " to set the proper permissions.

 
5-Load the module into Asterisk.
-Log into Asterisk’s CLI using " asterisk -r ".




-Load the module using " module load codec_g729.so " command.   





6-Verify the module has loaded correctly.
-Verify the codec has been loaded correctly using the "core show translation" command.
Great! you can see the codec at the end of the list.













7-Configure your phones to use G729 instead of choosing the G711a-law /G711u-law.
-Now,time to configure your phone to use G729.Connect to the FreePBX admin page,choose the extension that you want to configure it to use G729. disallow all codecs and allow only g729 to be used.


 8-Verifiy the codec usage by your phones on your Asterisk server.
-Make a call to any phone and use the " sip show channels " and congratulations!
here you can see the 2002 user is using G729 when placing or receiving calls.


Note : NOT all the softphone will support the G729 Codec.To download a softphone that can support G729 you can use SIPLite.Click here and after installing the softphone configure it to use only G729 from the codec settings as follows.



How To Solve The Problem Of The "Unavailable / Busy"message Recording ?!

Posted: Tuesday, May 14, 2013 by Unknown in Labels: , ,
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When using the *97 to access your voice-mail you can go to the advanced settings and record your "Unavailable / Busy" message but none of this messages will be played back to the caller when recording "Unavailable / Busy"message  using this way because *NOW has a bug that will prevent those messages from being played back to the caller if you are unavailable or busy see Ticket #1387 .you can solve this problem easily using the * feature code.

-What The * Feature Code Is Used For ?

1-To leave a voice mail message to any extension by directly dialing  the voice-mail of the user.for example if you want to leave a voice-mail message to extension 1001,you will dial * followed by the extension number like this *1001.

2-To access your voice-mail / the other users voice-mail by dialing the * followed by the extension number of the destination you want to access its voice-mail.for example if you want to access the voicemail of extention 1001 you will dial * followed by the extension number but you must interrupt the initial menu you will receive by pressing * and after a successful authentication you will log in and now you can record your "Unavailable / Busy"message. 

How To Solve The Problem Of The "ON-Demand" Recording?!

Posted: Monday, May 13, 2013 by Unknown in Labels: , ,
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On-Demand Call Recording Not Working?!Let's Fix It!:

By default,when enabling (Inbound / Outbound / Internal / External) call recording from the user extension configuration page , it will work without any problem but when enabling On-Demand recording it will fail so you need additional configuration steps. 

  1-Go To " FreePBX Advanced Settings "...Change The " Asterisk Dial Options ", " Asterisk Outbound Trunk Dial Options " As Follows :

Asterisk Dial Options : tTwWr
Asterisk Outbound Trunk Dial Options : TtwW

   -Setting the t allow the call recipient to transfer the call (*2 or ##).
   -Setting the T allow the call initiator to transfer the call (*2 or ##).
   -Setting the w allows the call recipient to toggle recording (*1).
   -Setting the W allows the call initiator to toggle call recording (*1).

2-Use " VI " To Edit The " features_featuremap_additional.conf " And  Add The " automon=*1 " :

vi /etc/asterisk/features_featuremap_additional.conf
automon=*1

3-Use " VI " To Edit The " features_general_custom.conf " And  Add The " featuredigittimeout=1500 " , " courtesytone=beep " :

vi  /etc/asterisk/features_general_custom.conf
featuredigittimeout=1500
courtesytone=beep

Note:You Must Reboot Your *NOW After Making The Changes To Take Effect.

AsteriskNOW / FreePBX Video Tutorial.

Posted: Friday, April 19, 2013 by Unknown in Labels: ,
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Asterisk makes it easy to build a variety of complex solutions including:
  •     IP PBX                                          
  •     IVR
  •     ACD
  •     VoIP Gateway
  •     Speech Applications
  •     Unified Communications
  •     Automated Attendant
  •     Voice Mail
  •     Unified Messaging
  •     Call Conferencing
  •     Call Recording

Download : AsteriskNOW 1

1.What is Asterisk?!                                     
-FreePBX : Call processing , phones features ( Call FW , Call Transfer , Call Pickup , On-Hold , etc ).     
-VM SRV : Your Voice Mail SRV.                                                                             
Download : http://www.asterisk.org/downloads
2.The Installation process.

Download : AsteriskNOW 2

1-The First Step After The Insallation?!Configure Static IP Configuration Using The "SETUP" Command.
2-How To Manage Your Asterisk SRV(GUI/CLI).
-GUI:The Web Administration Page .
FreePBX URL : http://192.168.1.9/admin
The Default Pass:admin  The Default Username:admin
VM SRV URL : http://192.168.1.9/recordings/
-CLI:The Asterisk Console , Asterisk CLI Module , SSH Using PUTTY , Java SSH Module

Download : AsteriskNOW 3

1-Time To Have Fun?! Time To Register Some Phones.
-What Phones?
-Physical / Softphones
-What Protocols?
-SIP VS IAX
-SIP Phones : X-lite , 3CX , For Windows and Ekiga , Jitsi For Linux.
-IAX2 Phones :ZoipER
-What Mode ?
-Extention Mode (The Default ) VS Device and User Mode.
2-Bais Phone Call using RTP and Wireshark Hacking.

Download : AsteriskNOW 4 

1-Baisc Phone Features.
-Dial by Name (SIP Alias)  -Call Transfer  -Info Services  -DND  -Call Forwarding
2-Enabling and Configuring Users VM .
3-How To Access Your VM / The Others VM Using The Voice Mail & Recordings Web Page / The Phone.
4-VMBlast Group Configuration.

Download : AsteriskNOW 5 

1-Advanced Phone Features PT-1.
-Wakup Calls
-Paging and Intercom
-Camp-On
-Parking
-Follow Me ( Mini Ring Group configurable by the user ).
-Ring Groups
-Pickup Groups
-Phone Book

Download : AsteriskNOW 6

1-Advanced Phone Features PT-2.
-Inbound Call Simulation Using  7777 Feature Code
-Blacklist
-Set Caller ID
-Call secreening
-Call Back
-DISA
-DISA + CB

Download : AsteriskNOW 7  
Download : AsteriskNOW 7  Tools

1-Advanced Phone Features PT-3.
-VmX Locater(Mini IVR Configurable By The User)
-System Recordings: Direct Recording VS Pre-Recorded Material
     -Bit Resolution:16Bit   -Sampling Rate:8000HZ/PCM    -Audio Channels:Mono
     -Visit:www.online-convert.com
-Announcements
-IVR AKA Digital Receptionist / Auto Attendant

Download : AsteriskNOW 8 NEW
Download : AsteriskNOW 8  Tools

 1-Advanced Phone Features PT-4.
-Conferences
-How To Solve The Problem Of The "Unavailable / Busy"message Recording ?!
       -The * =*98+*97
-Directory (Dial-by-name)
-Call Recording
       -How To Access The Recordinges ?
       -CLI : /var/spool/asterisk/monitor
       -GUI : "Voicemail & Recordings (ARI)"
-How To Solve The Problem Of The "ON-Demand"Recording?!

Download : AsteriskNOW 9 NEW
Download : AsteriskNOW 9 Tools

-How To Connect Two *NOW SRVs Using SIP/IAX User&Peer/Friend Mode ?
    -The Trunk Protocols  To Configure ?
    -SIP Vs IAX
    -The Trunk Mode/Type To Configure ?
    -User&Peer Vs Friend:
    -User&peer : Two Accounts To Authenticate.
    -Friend : Only One Shared  Account To Authenticate.