asterisk fast start advance dcap training
Asterisk is the world most popular and widely adopted open source telephony platform. It enables businesses to reduce cost on IP telephony and contact centers with maximum ROI.
Asterisk is also known as Swiss army knife or Lego building block that means its more than just a PBX – (Private Branch Exchange). It is used to develop any kind of telephony application with easy interface to any third party application like CRM or ERP. Integrating telephony with CRM or ERP increases the employee productivity.
Further, Asterisk can be integrated to any RDBMS – (Relational Database) to build a self-service phone system that eliminates the human latency while delivering services via phone system.
Asterisk is also widely used in ITSP – (Internet Telephony Service Provider) as soft switch and connects millions of callers daily.
To be able to use Asterisk more productively. One should have good understanding of Asterisk architecture and its core. Gaining insight knowledge of asterisk with practical experience is not possible without quality professional training.
Keeping in view regional demand for quality professional training. DVCOM provides DIGIUM – (Creator of asterisk) authorize training program for Asterisk. The training program is authorized by DIGIUM and offers the Asterisk fast start and Asterisk advance training with dCAP (DIGIUM Certified Asterisk Professional) exam facility.
For more information on these training, please contact DVCOM Training Dept.
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.
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.
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.
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.
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.
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.