Transfer Call

Any valid phone number to which to redirect the call, for example,

+16315551212

The + format makes the number location independent. This should be used unless you know all of your VCSs are in the same location. If you are using a variable, add + to the value of the variable, for example, +$NumberDialed$.

Note

• The maximum outgoing call number or SIP URI is 370 characters.
• This can also be an IP address. For H.323 calls, prefix the number with TA:, for example, TA:208.203.191.74. For SIP calls, use a prefix in the format sip:DNIS@IP_ADDRESS, for example, sip:[email protected], where 6111 is the DNIS.

This transfer type causes the VCS to access a second line from the gateway and generate an outbound call to the phone number specified in the Phone Number property. It then bridges the original caller and the outbound call together.

This transfer type dials the specified digits. No hook flash or other signalling is issued, so the script remains in control of the call and can interact with the original caller after the called party hangs up.

Note

This information is shown only if the Source display property is missing.

When transferring a call, you can optionally set the source address (CallingPartyNumber) from the script you are using.

If this is left blank, the ANI for the gateway is used, if that is not set, the ANI for the tenant is used.

This can be either an H.323-ID or an E.164 number:

• H.323-ID takes the format NAME:ARNIE or NAME:$anyvariable$
• E.164 takes the format TEL:1234 or TEL:$anyvalidnumberorvariable$

All possible values for this property should be contained in a single string, such as: NAME:NJONES,NAME:FSCHMIDT,TEL:1234567

The source address can be specified as either an H.323-ID value denoted by the NAME: prefix, or as an E.164 address denoted by the TEL: prefix

You do not have to send both types of source address, but it is legal if needed. However, if not blank, the source address must contain one or the other prefix format.

What to display as a source. This property is also known as CallerID or CallingPartyName.

Bridges the call control and media stream (the individual calls become a single call). If either party disconnects, the call terminates.

Parties can communicate with each other, but if one party disconnects, call does not terminate.

Do not bridge the calls, use the Bridge Call function, instead. See Bridge Call.