Free Online Asterisk Training classes - Class 03

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

Latency/Jitter

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

numan@numankhan.com

Thank you for the reading  have good time! J

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). 

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
support@expert-voice.com

Free Online Asterisk Training Classes - Class 02

Free Online Asterisk Training Classes

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). 

Regards!

Numan Khan

Expert-Voice