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VoIP
VoIP, or Voice over Internet Protocol, enables telephone calls through existing internet connections
instead of going through the traditional PSTN (Public Switched Telephone Network). It is not only
cost-effective, especially for a long-distance call, but also top quality voice calls over the internet.
Five sub-items to be covered to configure the VoIP feature, namely
SIP Device
,
Service Provider
,
SIP Account
,
Call Forward
,
Call Trough
,
Call Block
,
VoIP Dial Plan
,
PSTN Dial Plan, Phone
Book
SIP Device
Locale:
This selection is a drop-down box, which allows users to select the country for which the
VoIP device is operating. When a country is selected, the country parameters are automatically
loaded. Different countries can have their special ring mechanism.
SIP Port:
Set the SIP port, default 5060.
Dial Plan Priority:
Three modes for users to set the dial mechanism, default is set to Auto, thus
PSTN only with exception.
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Mode 0:
VoIP only and ignore all PSTN dial plans, send all calls to VoIP, including
Emergency calls.
Mode 1:
Default, which means that under this mode, the dial mechanism always match PSTN
plan first, then move to VoIP plan.
Auto:
Auto, this means the dial system will fall back to Mode 0 (VoIP) when no PSTN is
connected.
Quality of Service
User can mark DSCP for outgoing SIP and RTP. VoIP flow to control VoIP QoS.
DSCP Marking For SIP:
Set the DSCP marking for SIP VoIP packets for QoS proceeding.
DSCP Marking For RTP:
Set the DSCP marking for RTP VoIP packets for QoS proceeding.
T.38
T.38 relay is a way to permit faxes to be transported across IP networks between existing fax
terminals. The T.38 fax gateway converts and encapsulates the fax sent from the terminal fax
machines into a T.38 date stream. Then the gateway send the converted date packets to a T.38
enabled end point such as a fax or fax server or another T.38 gateway that converts it back to the
analog signal to realize the communication between two fax terminals.
T.38 Relay:
Click Enable to allow transmission of fax over IP network between two fax machines. If
T.38 relay is disabled, the analog fax signal is transmitted as the normal audio data. If T.38 relay is
enabled, the fax signal is converted to T.38 signal.
FAX Recipient’s path:
Set the path directly for storing the fax file to the storage.
Note:
For common fax usage, user should have a fax connected to the router, creating a fax
environment between two fax terminals, and the fax file(s) would be received through fax connected
to the router as what we usually perform.
But if user does not get a fax or he wants to store the fax to the file directly, he then can enable Fax
Reception feature. Select or enter manually the reception path for the file. (Here user can turn to
USB
for help.)
1) Set the field “Incoming Phone Port” to “FAX Reception” at the “VoIP Account” page.
2) Set the path user wants fax file saved at “FAX Recipient’s path” at the “SIP Device” page.
3) The incoming VoIP call for the specified VoIP account will be treated as Fax and saved to path.
FAX Recipient’s E-mail:
Enter the recipient’s email address. Once the fax file is delivered, the fax
file will be mailed to the account specified by the “Recipient’s Email”,
Delete Files After Sending:
The files will be deleted from system once the mail is sent out.
Answering Machine
The answering machine is a device for answering telephones and recording callers’ messages and
being enabled for both VoIP and PSTN.
The operation for the answering machine:
*#00:
Record user own greeting message;
1)
Start the recording after the beep sound
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2) Press # while finished.
3)
Hang up after the beep is heard. (system needs time on file translation and save to storage).
*#99:
Delete the user's greeting message
*#98:
Play the greeting message
*#xx:
Access the specified answering machine where xx (automatically designated by the system)
can be found at the "SIP Account" page.
*#96:
Enable the answering machine
*#97:
Disable the answering machine
1)
After the beep sound, dial the specified code xx where xx can be found at the "SIP Account"
page.
2)
Hang up after the beep is heard. (system needs time on file translation and save to storage).
*#90:
Access the PSTN A/M.
Note:
7800VDP(O)X uses the 1st available phone port to record the PSTN message. So, the answer
machine stops recording if user picks up the specified phone.
Greeting Delay:
The parameter is used as a threshhold for the answer machine to automatically
answer and record the messgae. There are seven items marking 0, 5, 10, 15, 20, 25, 30 respectively.
For example, if set to 0s, when there is an incoming call, the answering machine will respond
immediately and record the message. And if it is set to 20s, then the call will keep ringing until time
out of 20s (without user picking up the phone) before it can respond and record the meassge.
PIN:
The set password (no exceeding 8 digits) for listening to message. The customer should press
the PIN number so as to listen to the message. Leave it empty, and user can listen to the message
without entering password first.
Recipient’s Email:
Enter the recipient’s email address. Once the voice message is left (answering
machine operation), the voice message will be mailed to the account specified by the “Recipient’s
Email”,
Deleting Messages After Sending:
The message will be deleted from system once the mail is sent
out.
Delete All Messages:
Press the “Delete All” button to delete all messages stored in the system all at
once.
DND
User can set the time period here, during which both incoming VoIP and PSTN calls will be rejected.
Time Schedule:
When set to “Always On”, all the incoming calls will be rejected constantly; and also
you can set the precise time when the incoming calls are supposed to rejected. Or you can select
the already set timeslot in “
Time Schedule
”. And when set to “Disable”, there will be no time
restrictions on incoming calls. See
Time Schedule
.
Call Features
Call Wait:
Enable to activate Call Wait feature. When you are busy on a call, and another call comes
in, while the Call Wait feature is enabled, you can hear a hint sound indicating there is another call in
for you to decide whether to answer by slightly pressing the key “Flash” to keep the original call.
Three Way Conference Call:
Enable to activate three way conference call.
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Tone Control
Default Ring:
Support “Follow Locale selection” (Different countries have their special tone
mechanism) and another 4 embedded ring tones.
Dial Tone:
Support “Follow Locale selection” and another 5 embedded dial tones.
Gain Control
Gain control is to reduce the bad performance of quality issue caused by noise or echo, etc. Rx
means the performance of receiving and the Tx implies the performance of transimitting. A plus
quantity is to raise the performance while a negative quantity is to cut the performance (Rx: +1 to
increase the performance of receiving by 1 point and if set -1, the performance will be cut by 1 point,
the range is -20- 20.
).
PSTN Gain:
Set the PSTN gain, Tune the gain between -20-20 of the Rx and Tx respectively to
obtain a appropriate PSTN call environment.
Phone Port 1 Gain:
Set the gain. Tune the gain between -20-20 of the Rx and Tx respectively to
ensure a clear phone call.
Phone Port 2 Gain:
Set the gain. Tune the gain between -20-20 of the Rx and Tx respectively to
ensure a clear phone call.
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Service Provider
Register to a SIP service provider is an essential step before making the VoIP call. Users can find
out SIP service provider, and register a SIP account, jotting down the registration information and
configuring in router.
BiPAC 7800VDP(O)X offers a defaultSP item, you can change the settings or add a new Service
Provider yourself.
Service Provider Name:
Name of provider of the VoIP service
SIP Domain Name:
Enter the SIP registrar domain name.
SIP Proxy:
Also seen as SIP server, it manages the setup of calls between SIP devices including
the controlling of call routing and some necessary functions such as registration, authentication, and
network access control. Type the SIP Proxy address you obtain after you register from the service
provider.
SIP Proxy Port:
The port number set on your SIP proxy serve that the SIP proxy server uses to
make network connections, default is 5060.
SIP Outbound Proxy:
SIP outbound proxy is in similar use as SIP proxy, but when the SIP devices
are behind a firewall or a router or NAT, the SIP outbound proxy is the useful way to let SIP traffic to
pass from the internal network to the internet. Enter the SIP outbound proxy server address here.
SIP Outbound Proxy port:
Enter the port, normally 5060.
SIP Registrar:
Type the VoIP SIP registrar IP address.
SIP Registrar Port:
Type the port; it will listen to register requests from VoIP devices.

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