Admin Guide - Mizu Voip

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 deviceshave problems with call-transfer, the DTMF mode transfer is supported. If the transfer fails, than the Cisco “Also” method is also supported for transferfailover.-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 destination3: 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 eptransferstate; //0=unknown, 1 = mute to all, 2=talk with old client, 3 = talk with new client, 4 = talk with both clientsbecause we have 2 client ep (called) for the server (caller):GetOldTransferClient(); //will return the operatorGetNewTransferClient(); //will return the new clientWorkflow:Unconditional dtmf received (*9*number#)Set Istransferrer = trueRouteCall()If fail than play transferfail4.11.8. Three-Way CallingThree way dialogs with call holds are supported.