Tuesday, October 9, 2012

What you need to look before you deploy the VoIP in your organization.

The organization(s) that migrate from legacy PABX system (traditional phone system) to IP based PBX (Private Branch eXchange)   requires a very carefully planning before they deploy the VoIP.
In this article I am trying to explain that what exactly you need or the steps to ensure the QoS (Quality of Service) for VoIP in your organization.

Step 1
Analyze your existing LAN (Local Area Network) and find if there is any congestions /delays or unnecessary broadcasting or bursting.
The IP address that you plan to assign to the sip server ping it from several points/node in a network and find the ping statistics and you should look for
High latency/jitter will cause a very bad effect on VoIP quality and you will not be able to hear clearly.

Step 2
Check your network switches and theirs interfaces. All network switches need to have same interface capacity. It should not be the case that one switch you have is 1000 Mbps and the next switch is or after a next you have 100 Mbps.
This will cause network bottleneck.

Step 3
Create the VLAN (Virtual Private Network) to separate the data and voice traffic this will give a very good mileage to your voice traffic as they have separate network and you will have greater quality.

Step 4
If you don’t afford the VLAN (as it’s require manageable switch and relatively expensive as compare to non-manageable switch) then you should deploy all the 1000 Mbps switches in your network that will give more local bandwidth to your voice traffic.

Step 5
If you require the HD (High Definition) audio quality then install g722 audio codec. This is an optional but it will play a critical role in conference call or phone meeting as you will be able to hear clearly over the phone.

I hope this article has helped you to some extent. If you have any specific questions do write it to me at
Thank you for the reading  have good time! J

Sunday, July 1, 2012

SSCA SIP School Certified Associate

The SSCA training @ SIP School offer a very details and comprehensive training on SIP. It provides a solid background on this emerging technology and cover in depths topics on different aspects of SIP (Audio/Video/IM/Fax).
I have privileged to be a SSCA (Verification ID: 1340896947997-08378). 

Tuesday, June 5, 2012

Integrate Asterisk with Zoho CRM

The Asterisk (trixbox/elastix) is able to integrate with Zoho CRM (http://www.zoho.com) using PhonebBridge application developed by Zoho Team.
It requires the JDK (Java Development Kit) to be installed on Linux machine and configure the AMI (Asterisk Manager Interface) to work.
It helps to call directly from the Zoho CRM to any phone number and can save the call logs /description.
Very useful for the CRM (Customer Relationship Management) users to save a lot of time when making calls to the client/customers.
For professional support on this Zoho Phone Bridge. Please contact

Wednesday, April 11, 2012

Difference between Asterisk trixbox Elastix AsteriskNow PBX in Flash freepbx

As a Asterisk professional I came across several people asking for the difference between asterisk, trixbox, Elastix, AsteriskNow, PBX in Flash or even FreePBX.
So the answer is there is no difference at all. Each one of these above mentioned software are running the Asterisk as core telephony engine and front end as FreePBX  moreover trixbox and Elastix has modified the look and feel of free open source available freePBX. (http://www.freepbx.org) and has created their own version of front end tool. 
now the question which one is best to use, so the best and trouble free is plain asterisk that you download from asterisk.org and compiled it  as its exactly contains what you need and you don't get unnecessary software packages and binaries like in case of trixbox ce and elastix where you get many un-used packages that you don't even requires in some cases for example MySQL, Apache, extra Sound files and there are many more. 
Moreover my experience with trixbox is, its mainly run several AGI (Asterisk Gateway Interface) scripts that are developed in PERL language and if overloaded it crashes frequently and on main dashboard it also show you wrong indicator of CPU usage and you keep on restarting your server (in case if you are newbie) where actually there is no such issue with CPU load.

To have a survival skills to run asterisk base system you need to be average computer programmer because when you design your dialplan (heart of asterisk). It has its own scripting like language and have computer programming like language constructs (variables, expressions, if -if then else statement, sub-routine) etc.

other then this you also need to be good Linux administrator that can perform basic task as well.
Asterisk is daemon service that run on Linux so to run the service you need to be able to perform operating system maintenance task. 

If you are working for a call center/customer contact center, you must have also RDBMS skills as well to maintain the CDR (Call Detail Report). 

Numan Khan

Sunday, March 4, 2012

Install g729 codec free

The g729 codecs are the most popular codes for VoIP particularly when using a VoIP trunk for a distance calling and you don't have enough bandwidth to make a crystal clear and high definition audio call. 
I will be showing you for the simple way to install the g729 code as free for you Asterisk box but before we proceed please take a note for the following. 

The free g729 codes are only supported in Asterisk for the multiple channels as 'Pass through mode' .
What does it mean ? 
The 'Pass through mode' means that your call originating device & call termination devices must have support for g729. Otherwise it will show you Error codecs translation and you can't play IVR using g729 sound files format and also can not do the call recordings with g729 as recorded file format. If you want to have calling recording or IVR then you must purchase the license for g729 and its per port means per channel more information can be found on Digium website as follow. 

Since we don't need call recording or IVR to be played using g729 codecs format. and we require only to use this for making calls and less consume our bandwidth. 

Following are the steps 
1. Find the processors type of your machine by using the command

 uname -a

cat /proc/cpuinfo

in my case its i386 and CPU is Pentium 4 and then further find your Asterisk version by using Asterisk  -V
and again my version of  asterisk is its fonality compiled means its Trixbox 2.8.

now we have Processor type as i386 and family of Pentium 4 with Asterisk with these information we are going to download the g729 codes for our asterisk box. 

as you can see in the above screen I have selected asterisk 1.6.0 branch and if you further scroll down you will find the following. 

This is what we need right click on it and select the 'copy link address' (Firefox & Chrome) users 
and now go to your CLI terminal and navigate to the following directory. 
using cd command. 

cd /usr/lib/asterisk/modules

and type
wget http://asterisk.hosting.lv/bin160/codec_g729-ast16-icc-glibc-pentium4.so

and hit enter it will start downloading the codecs. 
after it successfully download issue the command

ls -ltr | grep g729

you will see the files with .so as extension 
check if you have any old version of g729 codes just delete it by using
rm -rf codecs_g729.so 

Please be care full here there is file name 'format_g729.so' (DON'T DELETE THIS ONE)

and now just renamed the downloaded codes as follow

mv codec_g729-ast16-icc-glibc-pentium4.so codec_g729.so

and check again 

ls -ltr | grep g729

you will have 

-rwxrwxrwx 1 root      root      397024 Sep 17  2008 codec_g729.so
-rwxr-xr-x    1 root      root        7124 Jun  8  2010 format_g729.so

its better to allow permission rights on this codec so I am giving full right permission 

chmod 777 codec_g729.so

and its time to restart the asterisk

/etc/init.d/asterisk restart

Note: If asterisk keep restarting with some strange behavior. Its better to reboot this machine and after rebooting connect to Asterisk CLI using 
Asterisk -r and type core show translation
you will see the g729 codes along with others code as I highlighted in red. If this column and rows shows numeric values means you have install it successfully otherwise repeat all the steps and try again. 

Next use allow g729 codes in your sip.conf and enjoy free this great quality codes for free. Make some calls and you will notice the difference the calls are more clear and audio able then before if you are using g729 for the first time. 

Thanks for reading
Numan Khan
Expert Voice 

Saturday, February 11, 2012

Connect Asterisk with Nokia E63

I have been figuring out to connect my cell phone Nokia E63 to my asterisk box  over a wifi network and finally I got it actually its not as hard just follow the steps below
If you have Nokia E63

go to the Tool>>Setting>>Connection>>Sip Settings

Chose New SIP Profile.
Profile Name =  name it anything as you wish
Service Profile = IETF
Default Access point = choose your wifi connection
Public User name = asterisk sip extension, in my case it is 200
Use Compression = No
Registration = Always on
Use Security = No
Proxy Server:
Proxy Server Address: Your Asterisk Machine IP address
username= your sip extension in asterisk
Password= your sip extension password
Allow loose routing = Yes
Transport Type = Auto
Port = 5060
Register Server
Register Server address = your asterisk machine IP

username= your sip extension in asterisk
Password= your sip extension password
Transport Type = Auto
Once you finished all these setting press back and on SIP setting menu you'd see the status as
Registered (Wifi Network only)

now to make a call you need to do little things more,

go to Tools>>Settings>Connections>>Internet tel.

and create a new profile and select this SIP Profile and press back

The great thing about Nokia E63 is it can automatically switch between voice GSM mode and Internet mode. This feature may also be there in others Nokia model but I only know about E63.
Now to get this go to the
Default call type=Internet call

and in the Network option select dual mode
Now if I am in office my cell phone act as my local extensions and I send receive calls using landline and If I am at home I am connected to my SIP server and can make a cheap/almost free calls internationally.

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