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Vigor2930 Series User’s Guide
178
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.
Call Forwarding
There are four options for you to choose.
Disable
is to close
call forwarding function.
Always
means all the incoming calls
from Internet 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., [email protected] or
[email protected]) 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
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igor2930 Series User’s Guide
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detailed configuration.
Index (1-60) in Phone Book -
Enter the index of phone book
profiles. Refer to section
3.11.1 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.
Voice Active Detector -
This function can detect if the voice
on both sides is active or not. If not, the router will do
something to save the bandwidth for other using. Click On to
invoke this function; click off to close the function.
Default SIP Account
You can set SIP accounts (up to six groups) on SIP Account
page. Use the drop down list to choose one of the profile
names for the accounts as the default one for this phone
setting.
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Play dial tone only when account registered -
Check this
box to invoke the function.
Default Call Route
It determines the default direction for the call route of the
router.
To ISDN (for VoIP)
- The router is set by using ISDN call.
To change ISDN call into VoIP call, please dial the character
in this field for transferring. The character that you can type
can be *, #, and 0~9.
To VoIP (for ISDN)
- The router is set by using VoIP call. To
change VoIP call into ISDN call, please dial the character in
this field for transferring. The character that you can type can
be *, #, and 0~9.
In addition, you can press the
Advanced
button to configure tone settings, volume gain,
MISC and DTMF mode.
Advanced
setting is provided for fitting the telecommunication
custom for the local area of the router installed. Wrong tone settings might cause
inconvenience for users. To set the sound pattern of the phone set, simply choose a proper
region to let the system find out the preset tone settings and caller ID type automatically. Or
you can adjust tone settings manually if you choose User Defined. TOn1, TOff1, TOn2 and
TOff2 mean the cadence of the tone pattern. TOn1 and TOn2 represent sound-on; TOff1 and
TOff2 represent the sound-off.
Region
Select the proper region which you are located. The common
settings of
Caller ID Type
,
Dial tone
,
Ringing tone
,
Busy
tone
and
Congestion tone
will be shown automatically on the
page. If you cannot find out a suitable one, please choose
User Defined
and fill out the corresponding values for dial
tone, ringing tone, busy tone, congestion tone by yourself for
VoIP phone.
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181
Also, you can specify each field for your necessity. It is
recommended for you to use the default settings for VoIP
communication.
Volume Gain
Mic Gain (1-10)/Speaker Gain (1-10)
- Adjust the volume of
microphone and speaker by entering number from 1- 10. The
larger of the number, the louder the volume is.
MISC
Dial Tone Power Leve
l - This setting is used to adjust the
loudness of the dial tone. The smaller the number is, the
louder the dial tone is.
It is recommended for you to use the
default setting.
Ring Frequency
- This setting is used to drive the frequency
of the ring tone. It is recommended for you to use the default
setting.
DTMF
DTMF Mode –
There are four DTMF modes for you to
choose.
InBand
-
Choose this one then the Vigor will send the DTMF
tone as audio directly when you press the keypad on the phone
OutBand
-
Choose this one then the Vigor will capture the
keypad number you pressed and transform it to digital form
then send to the other side; the receiver will generate the tone
according to the digital form it receive. This function is very
useful when the network traffic congestion occurs and it still
can remain the accuracy of DTMF tone.
SIP INFO
-
Choose this one then the Vigor will capture the
DTMF tone and transfer it into SIP form. Then it will be sent
to the remote end with SIP message.
Payload Type (rfc2833)
- Choose a number from 96 to 127,
the default value was 101. This setting is available for the
OutBand (RFC2833) mode.
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Detailed Settings for ISDN1-S0 Port
Click the number link of Index 3 (ISDN1-S0), you can access into the following page for
configuring Phone settings.
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.
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., [email protected] or

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