AX-125/145 SIP VoIP Router Quick Installation Guide
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•
If the relative setting fields are left blank, AX-125/145 will disable this function.
You can predefine all the call service command in the web GUI in advance, or
just use the default settings. Refer to
Table 2. Default Call Functions
Settings
for more details.
•
Unless there is any conflict, ASUS strongly recommend that you should not
change the default service codes.
• If AX-125/145 successfully registered to the SIP server, then you will hear
a dial tone when you pick up the phone set, and you will hear a busy tone if
AX-125/145 failed to register. You can still operate some functions when you
hear a busy tone, such as IVR setting,or phone book dialing.
4.3 PSTN Access
By default, the number that you dialed on the phone set is interpreted as a VoIP
call and is sent to the SIP server.
If you have connected the FXO to the PSTN line of the AX-125/145, then you can
use the PSTN access code. The default code is "
*90
" and this is configurable in the
web GUI.
To transfer a call
1.
Blind Transfer
: Transferring a call to a third party without notifying the
recipient. Press the “
Flash
” key and dial “*
98nnnn#
” (nnnn is the recipient''s
number). Then, put down the phone.
2.
Attend Transfer
: Transferring a call to a recipient and ensuring the call is
successfully transferred. Press the "
Flash
" key and dial “*
88nnnnn#
” (nnnnn
is the recipient's destination number). When the recipient answers, put down
the phone.
To make a three-way conference call
If you want to invite a third party to your current call, press the "
Flash
" key and
dial *
89nnnnn#
” (nnnnn is the recipient's destination number). When the recipient
answers, press the "Flash" key.
To Pick up a call
Dial “*
78nnnnn”
(nnnnn is the number of the ringing phone) to answer an incoming
call.
