Master Microsoft Teams Voice - Phone System - Direct Routing

Learn Microsoft Teams Direct Routing for On-Premise PBX Integration with Certified SBC.

• Do you wish to migrate PBX users from on-premises to the cloud?

• Do you want to keep or need to keep your current on-premises legacy PBX as a hybrid system to use loud UC as well?

• Does your current legacy PBX offer unique features that are critical to your business?

• Do some or all of your users need features that the Phone System does not currently provide?

Microsoft's Phone System technology allows Microsoft Teams to use PBX and call control features in the Microsoft 365 cloud.

The phone system is compatible with MS Teams clients and approved gadgets. You can use Phone System to replace your current PBX system with a set of services that are directly provided by Microsoft 365.

Internal calls within your company are handled only within the Phone System and never across the Public Switched Telephone Network (PSTN), eliminating the need for long-distance charges.

The phone system offers add-on options for connecting to the PSTN in order to place external calls.

You can link a supported, customer-provided Session Border Controller (SBC) to a phone system using Direct Routing.

Using this feature, you can set up on-premises Public Switched Telephone Network (PSTN) connectivity with the Microsoft Teams client.

In this course, students will learn to configure the Microsoft Phone system by following hands-on labs as demonstrated in the course. The course teaches all the advanced features offered by Microsoft Teams Enterprise Voice.

All lab exercises use the Microsoft Azure Cloud platform. Students will be able to configure the phone system direct routing step by step by installing MS Teams-certified SBC and trunking with Asterisk PBX for the demonstration.

for complete course, visit - 

https://www.udemy.com/course/master-microsoft-teams-voice-phone-system-direct-routing

Asterisk-Elastix VoIP - Learn to Build Phone System

Learn Elastix IP Telephony server configurations and deployment with course project.

Course Description

• Do you need a cost effective phone system for business?
• Are you tired of paying licensing cost to the proprietary phone system ?
• Do you want to expand your business and require phone system but legacy system don't allow you ?
• You are interested to learn asterisk but like to avoid command line and linux shell at the start ?

Elastix is the world most popular and widely adopted open source IP telephony software. The core VoIP communication is based on Asterisk - The most powerful IP telephony platform. This course is designed for the newbies, small & medium business that like to use the IP telephony - PBX or even the solution providers that like to gear up for telephony services to the end users.

The course starts with initial telephony concepts and terms used in phone service industry. No prior technical knowledge about telecommunication required to take this course.

The course has project depicting the real world scenario. This helps the students to have activity based learning and can apply learned knowledge to the real world. The course clearly explained the phone system features with lab practice for configurations.

Proprietary systems invites a license fee for most Business-critical features and UC Features like mobility and third party integration. To avoid the expensive add-ons go the open source way.

Use a single system to manage multiple tools and platforms, thereby having “greater employee productivity, reduced costs and a means to improve customer engagement.

What are the requirements?

• Download the Elastix software but don't we will cover in Installation lecture.
• Download Softphone e.g Zoiper or Xlite. Will show you configuration in lecture 04.

For the complete course details please visit the below link. 

Elastix IP Telephony Complete Video Training. 

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