Unicorn 3112 User Manual
Copyright © 2008
Hanlong Technology Co., Ltd
Page 11 of 42
second NOTIFY signal from the transferee and decided to time out.
Note: this does not indicate the transfer has been successful, nor does it indicate the
transfer has failed. When transferee uses a device that does not support the second
NOTIFY signal, this will be the case. In poor or unstable network scenarios, this could also
happen, although the transfer may have been completed successfully.
5.2.7. Attended Transfer
Assuming that call party A and party B are in conversation. Party A wants to Attend
Transfer party B to party C:
Party A presses FLASH (on the analog phone, or Hook Flash for old model phones) to
get a dial tone.
Party A then dials party C’s number then # (or wait for 4 seconds). Party A and party C
now are in conversation.
Party A can hang
Note: When Attended Transfer failed and if party A hangs up, the Unicorn 3112 will ring
party A again to remind party A that party B is still on the call, by pressing FLASH or Hook
again will restore the conversation between party A and party B.
5.2.8. Send and Receive PSTN Calls
Users can send and receive calls from PSTN. To receive PSTN calls, simply take the
phone off hook when the analog phone rings. To make a PSTN call, first press *00 (or
your own PSTN Access Code) to get the PSTN line dial tone and dial the PSTN number.
5.2.9. VoIP-to-PSTN Calls
To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone number first.
A ring tone is played once followed by a dial tone. At this time, users can dial a PSTN
telephone number or a mobile telephone number then # (or wait for 4 seconds). The call
will be established afterwards. If no PSTN number is entered after the dial tone, Unicorn
3112 will hang up automatically in 10 seconds.
In the web configuration page, if the Route to PSTN field is configured, the second stage
dialing is eliminated. That is, after users dial the FXO SIP account number, the PSTN
number will be called automatically.
5.2.10. PSTN-to-VoIP Calls
To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the FXO port
