Business Phone Number ​

In practical terms, a DID (Direct Inward Dialing) Number or a Virtual Number (also called a DDI in Europe) is a local telephone number in a selected country or city. The calls made to or made from these numbers are then forwarded by CCS to Voice over IP (VoIP) using SIP, H.323 or IAX protocols, or to a PSTN (Plain Old Telephone System) device such as a regular or mobile telephone anywhere in the world.

Typically, a DID Number is used to provide local telephone numbers for customers on a worldwide basis, or to provide access from regular telephones to non-PSTN destinations such as VoIP.

As your new DID number is based on the cloud, getting it set up is almost instant. Once numbers are demanded, your numbers can be activated and ready to go in a very short time depending on country specific regulations.

Call Center Studio supports international calling to every single country in the world. For local outbound calling please talk to an expert since it depends on country specific regulations.

A SIP trunk is the modern-day equivalent of a T1 trunk. Where in the past you would buy a T1 trunk from a Telco provider and connect it to your legacy PBX, today you can make a purchase from an ITSP (Internet Telephony Service Provider) and connect it to your VoIP / IP PBX. SIP trunks use the SIP standard. The name ‘trunk’ comes from the telecom world and it means, more or less, a group of phone lines.

Using this standard protocol, telecom service (VoIP) providers connect one or more channels to the customer’s PBX. Phone numbers and DIDs are linked to the SIP trunk and in many cases, numbers can be ported to the new service.

The terms IP PBX and VoIP phone system are often used interchangeably to describe a business phone system that uses Internet Protocol (IP) to transmit voice. VoIP therefore simply stands for ‘Voice over Internet Protocol’. The switchboard element of your private phone system is the PBX, which stands for private branch exchange.

An IP PBX system is the central component of most modern VoIP phone systems. These systems consist of the IP PBX server, VoIP endpoints (i.e fixed phones, web clients, or mobile apps), and optionally, a VoIP Gateway. The IP PBX server is similar to a proxy server. SIP clients, (either soft phones or hardware-based phones), register with the server, and when they wish to make phone calls they ask the IP PBX to establish the connection.

IP PBXs have a directory of all phones/users and their corresponding SIP addresses. This enables them to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider to the desired destination.

There are fourteen SIP Request methods of which the first six are the most basic request / method types:

INVITE = Establishes a session
ACK = Confirms an INVITE request
BYE = Ends a session
CANCEL = Cancels establishing of a session
REGISTER = Communicates user location (host name, IP)
OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones
PRACK = Provisional Acknowledgement
SUBSCRIBE = Subscribes for Notification from the notifier
NOTIFY = Notifies the subscriber of a new event
PUBLISH = Publishes an event to the Server
INFO = Sends mid session information
REFER = Asks the recipient to issue call transfer
MESSAGE = Transports Instant Messages
UPDATE = Modifies the state of a session

SIP Requests are answered with SIP responses, of which there are six classes:

1xx = Informational responses, such as 180 (ringing)
2xx = Success responses
3xx = Redirection responses
4xx = Request failures
5xx = Server errors
6xx = Global failures