Draytek Vigor 2110, Руководство пользователя

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Vigor2110 Series User’s Guide 164
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
Application>>Schedule for
detailed configuration.
Index (1-60) in Phone Book - Enter the index of phone book
profiles. Refer to section DialPlan>> Phone Book for detailed
configuration.
CLIR (hide caller ID) Check this box to hide the caller ID on the display panel of the
phone set.
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.
Single Codec – If the box is checked, only the selected Codec
will be applied.
Packet Size -The amount of data contained in a single packet.
The default value is 20 ms, which means the data packet will
contain 20 ms voice information.