Attended Transfer explained
Can you think of a situation when someody calls a company, an operator answers instead of the right person, so that call needs to be transferred but that person needs to be asked first? Like in a call center? Do you need confirmation before you transfer a call to someone else? Then you need to use Attended Transfer. Read this article to find out why.
An attended transfer is a transfer made after notifying the destination party of the transfer, so a call is only routed to a third party if that third party answers the call. In other words when X transfers a call received from A to B after B has answered, like in a call center.
The call center agent who manages all the incoming calls has to choose which agent to transfer the call to, according to the caller's needs. Then he or she calls that agent to tell him or her about that specific call before actually performing the call transfer. It is the most useful to match the needs of the caller to the best suited agent.
It means that a call is made by an application and when it is ready to transfer then a REFER message (see SIP Basics explained) is sent to the caller, and it will contain two headers: TO and REPLACES, both to identify the third party. Then the caller sends the message INVITE to the third party, with the necessary REPLACES information. So, for attended transfer, both phones must support both REFER and REPLACES.
Figure 1 - Attended Transfer explained
Process of Attended Transfer (as demonstrated by Figure 1)
- #1 calls #2
- #2 answers and places #1 on "hold" by pressing the "Line 1" or "Line 2" button
- Now #2 dials #3, #3 answers.
- To transfer the call from #3 to #1, #2 presses the "transfer" key, followed by the "OK" key
- #2 is disconnected from #1 and #2
- #1 is transferred to #3
In the meantime, several other lines are maintained, and several other sessions are held on the same PBX (Private Branch eXchange handles all calls within a local network).
You need these before you start using the voip call (Voice over Internet Protocol is a protocol responsible for multimedia transmission over the Internet) transfer:
- A connection to SIP telephones within a SIP (Session Initiation Protocol, which is responsible for establishing, maintaining and terminating connection between two conversation partners in a multimedia session)network that support the REFER method with a REPLACES header.
- An application that performs a transfer operation with a screened transfer that has simple call-answer supervision.
- An imported custom server and state tables.
If we keep our example of a call center, the simplest solution would be to have a human operator to transfer the calls directly by user interaction. It would mean that the operator accepts the call, asks about the intention of the caller, then decides who to transfer the call, based on the caller's answers. Then he or she types in the number of the desired extension to transfer the call.
If the operator calls that extension before the transfer it would mean an attended transfer, while in case of a blind transfer he or she would not call the end-point before the transfer. In the former one the operator needs to perform two simultaneous calls, while in the latter one it is not needed.
The two simultaneous calls only exists while the operator calls the agent to whom the call is about to be transferred, and having got the agent's approval, then the operator chooses to transfer the call and leaves the conversation. Then one of the two simultaneous calls would be terminated, and only one line will be needed further on.
Attended Transfer is one of the great deal of features available at Ozeki Phone System. It is the only VoIP PBX that offers you multifuncionality, advanced technology, easy manageability and high expertise. If you need to expand your availability via communication technologies through VoIP, then the best choice as you may have realized (if you have not, I am sure you will soon) is Ozeki Phone System.
These articles have more information about certain topics. Read them to know more:
More information
- Voice Codecs explained
- SIP Trunking explained
- SIP Protocol explained
- RTP Protocol explained
- H.323 Protocol explained
- SIP Basics explained
- Dial Plans explained
- VoIP VLAN explained
- VoIP explained
- VoIP Security explained
- SIP explained
- Attended Transfer explained
- VoIP Routing explained
- VoIP Billing explained
- VoIP SIP Tutorial explained
- Phone Calls on the Internet explained
- Internet Calling explained
- VoIP Monitoring explained
- VoIP Calling explained
- PC to Phone Calls explained
- Call Termination explained
- GSM VoIP Gateway explained
- Fring VoIP explained