System information

4.11.7. Call Transfer
-The standard sip calltransfer protocol is supported (REFER, replaces methods) using the transfer button on sip phones, but because many sip devices
have problems with call-transfer, the DTMF mode transfer is supported. If the transfer fails, than the Cisco “Also” method is also supported for transfer
failover.
-Calls can be transferred by the following dtmf digits:
*9*number# -> unattended transfer
*8*number# -> transfer with consultation
-You will hear a music if the calltransfer is in progress, or a failure notice if it failed.
-After the transfer successfully begun (ringing to the new client), you have the following DTMF options:
1: talk with the new called party when it is connected (not possible if already is connected),
for example to discuss the purpose of the caller.
When you hang-up, the caller and the new called party will be connected.
You don’t need to press 1 if you have started the transfer with consultation (*8*).
2: disconnect the new called party and talk with the original caller (not possible if already transferred)
for example you can redirect the caller to a new destination
3: stay in line with the old client (you have to hang-up when you finish to complete the transfer)
Note: DTMF ** will always reset your already entered digits to *
Technical description:
Structures:
calltype = eTransfer
//for the transferrer client ep
transferstate; //0=unknown, 1 = mute to all, 2=talk with old client, 3 = talk with new client, 4 = talk with both clients
because we have 2 client ep (called) for the server (caller):
GetOldTransferClient(); //will return the operator
GetNewTransferClient(); //will return the new client
Workflow:
Unconditional dtmf received (*9*number#)
Set Istransferrer = true
RouteCall()
If fail than play transferfail
4.11.8. Three-Way Calling
Three way dialogs with call holds are supported.