Section 5.2. Basic Call Handling


5.2. Basic Call Handling

Call-handling features are the essential traffic-processing duties of a PBX and the group of applications that most resembles the services available to a POTS subscriber. Connecting, disconnecting, and call transfer are the building blocks of all call-handling applications.

5.2.1. Intercom Call

Intercom calling is the oldest of all telephony applications. In fact, it was the intercom calling application that lead to the invention of the telephone. This one telephony application came even before the network did. When Alexander Graham Bell's assistant heard the inventor 's voice coming through a transducer, quite by accident , the first intercom call was placed.

An intercom call is merely a conversation between two private endpoints. Most intercom calls are two-way conversations, meaning both parties can talk and hear each other talk. In PBX terms, intercom calls are calls between extensions. The setup in Project 4.2 allows you to place intercom calls between the two IP phones.

5.2.2. Mute, Hold, Call Transfer, and Multiparty Conference

Certain features have become more or less staples of a PBX environment. Mute allows the local party to temporarily stop sending sound to the remote party. Hold is a form of mute that allows the local party to leave the phone set for an extended period of time, unattended, and then return to resume the conversation with the remote party. Call transfer allows the local party to hand the call in progress off to another extension, so it can be resumed there. Multiparty conference allows the local party to set up conference calls from her phone.

5.2.2.1 Blind and consultative transfer

A blind call transfer is one in which the transferring party hangs up immediately following the transfer, having no knowledge of the availability of the party to whom the call was transferred. A consultative transfer allows the transferring party to speak with the receiving party prior to making the transfer or merely verify the receiver's availability before completing the transfer.

5.2.2.2 Conference

Multiparty conference , or just conference , allows more than two parties to have a call, all able to hear one another at the same time. With PBXs, these features are accomplished by proprietary programming. In VoIP environments, packet-based signaling systems like SIP and MEGACO (media gateway control protocol) work with DSP (digital signal processing) servers called conference mixers to facilitate this kind of functionality. In the PSTN, SS7 does the job.

5.2.2.3 Meet-me

Meet-me conferences are ad hoc conferences in which users can voluntarily join a conference in progress by dialing a code on their phones. This code can be permanently set in the PBX on behalf of a certain user , or it can be established on the fly by the host of the conference. Users attempting to join the conference may be required to enter a password or PIN code. Not all PBX systems support this kind of conference.

5.2.3. Caller ID

Caller identification , also called calling party identification , allows the recipient of a phone call to know which endpoint is calling both before and after answering. Caller ID signals can be sent using in-band or out-of- band signaling. Some endpoints are able to display the ID information on a built-in display. Some endpoints require an outboard display device. With CTI (computer-telephony integration) programming, caller ID information can be displayed as a part of a PC or web application. SoftPBXs can use caller ID data as criteria for call routing.



Switching to VoIP
Switching to VoIP
ISBN: 0596008686
EAN: 2147483647
Year: 2005
Pages: 172

Similar book on Amazon

flylib.com © 2008-2017.
If you may any questions please contact us: flylib@qtcs.net