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Vigor2910 Series User’s Guide
118
Hotline Check the box to enable it. Type in the SIP URL in the field
for dialing automatically when you pick up the phone set.
Session Timer Check the box to enable the function. In the limited time that
you set in this field, if there is no response, the connecting call
will be closed automatically.
T.38 Fax Function If the remote end also supports FAX function, you can check
this box to enable this function.
Call Forwarding There are four options for you to choose. Disable is to close
call forwarding function. Always means all the incoming calls
will be forwarded into SIP URL without any reason. Busy
means the incoming calls will be forwarded into SIP URL
only when the local system is busy. No answer means if the
incoming calls do not receive any response, they will be
forwarded to the SIP URL by the time out.
SIP URL – Type in the SIP URL (e.g., aaa@draytel.org or
abc@iptel.org) as the site for call forwarded.
Time Out – Set the time out for the call forwarding. The
default setting is 30 sec.
DND (Do Not Disturb)
mode
Set a period of peace time without disturbing by VoIP phone
call. During the period, the one who dial in will listen busy
tone, yet the local user will not listen any ring tone.
Index (1-15) in Schedule - Enter the index of schedule
profiles to control the DND mode according to the
preconfigured schedules. Refer to section 3.5.2 Schedule for
detailed configuration.
Index (1-60) in Phone Book - Enter the index of phone book
profiles. Refer to section 3.10.1 DialPlan – Phone Book for
detailed configuration.
Call Waiting Check this box to invoke this function. A notice sound will
appear to tell the user new phone call is waiting for your
response. Click hook flash to pick up the waiting phone call.
Call Transfer Check this box to invoke this function. Click hook flash to
initiate another phone call. When the phone call connection
succeeds, hang up the phone. The other two sides can
communicate, then.
Prefer Codec Select one of five codecs as the default for your VoIP calls.
The codec used for each call will be negotiated with the peer
party before each session, and so may not be your default
choice. The default codec is G.729A/B; it occupies little
bandwidth while maintaining good voice quality.
If your upstream speed is only 64Kbps, do not use G.711
codec. It is better for you to have at least 256Kbps upstream if
you would like to use G.711.