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Ring Pattern
Choose a ring tone type for the VoIP phone call.
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.
4.12.3 Phone Settings
This page allows user to set phone settings for Phone 1 and Phone 2 respectively. However, it
changes slightly according to different model you have.
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Phone List
Port –
there are two phone ports provided here for you to
configure.
Phone1/Phone2
allowS you to set general settings
for PSTN phones.
Call Feature
– A brief description for call feature will be
shown in this field for your reference.
Codec
– The default Codec setting for each port will be shown
in this field for your reference. You can click the number below
the Index field to change it for each phone port.
Tone
- Display the tone settings that configured in the advanced
settings page of Phone Index.
Gain
- Display the volume gain settings for Mic/Speaker that
configured in the advanced settings page of Phone Index.
Default SIP Account
– “draytel_1” is the default SIP account.
You can click the number below the Index field to change SIP
account for each phone port.
DTMF Relay
– Display DTMF mode that configured in the
advanced settings page of Phone Index.
RTP
Symmetric RTP
– Check this box to invoke the function. To
make the data transmission going through on both ends of local
router and remote router not misleading due to IP lost (for
example, sending data from the public IP of remote router to the
private IP of local router), you can check this box to solve this
problem.
Dynamic RTP Port Start
- Specifies the start port for RTP
stream. The default value is 10050.
Dynamic RTP Port End
- Specifies the end port for RTP
stream. The default value is 15000.
RTP TOS
– It decides the level of VoIP package. Use the drop
down list to choose any one of them.
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Detailed Settings for Phone Port
Click the number link for Phone port, 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
[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)
Set a period of peace time without disturbing by VoIP phone
call. During the period, the one who dial in will listen busy
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mode
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 call barring 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.
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.
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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.
Play dial tone only when account registered -
Check this box
to invoke the function.
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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