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The major benefit of this mode is that you don’t have to memorize your friend’s IP
address, which might change very frequently if it’s dynamic. Instead of that, you will
only have to use
dial plan
or directly dial your friend’s
account name
if you are with
the same SIP Registrar.
Peer-to-Peer
Before calling, you have to know your friend’s IP Address. The Vigor VoIP Routers will
build connection between each other.
Our Vigor V models firstly apply efficient codecs designed to make the best use of
available bandwidth, but Vigor V models also equip with automatic QoS assurance.
QoS Assurance assists to assign high priority to voice traffic via Internet. You will
always have the required inbound and outbound bandwidth that is prioritized exclusively
for Voice traffic over Internet but you just get your data a little slower and it is tolerable
for data traffic.
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3.15.1 DialPlan
This page allows you to set phone book, digit map, call barring, regional settings and PSTN
setup for the VoIP function. Click the links on this page to access into next pages for detailed
settings.
Available settings are explained as follows:
Item
Description
Enable Secure Phone
It allows users to have encrypted RTP stream with the peer
side using the same protocol (ZRTP+SRTP). Check this
box to have secure call.
Enable SAS Voice Prompt
If it is enabled, SAS prompt will be heard for both ends
every time. If it is disabled, no SAS prompt will be heard
any more.
Application for Secure Phone
Enable SAS Voice Prompt, for ex: if vigor router A calls vigor router B with checking
Enable
Secure Phone
and
Enable SAS Voice Prompt
, then:
1.
After the connection established, vigor router A will send SAS voice prompt to A and
vigor router B will send the SAS voice prompt to B.
2.
Then the RTP traffic is secured until the call ends.
3.
If vigor router A wants to call vigor router B again next time, both A and B will not hear
any voice prompt again even checking
Enable SAS Voice Prompt
on web UI. It means
only the first call between them will have voice prompt.
Enable SAS Voice Prompt, for ex: if vigor router A calls vigor router B with checking
Enable
Secure Phone
but not
Enable SAS Voice Prompt
, then:
1.
After the connection established, vigor router A will
NOT
send SAS voice prompt to
vigor router A and vigor router B will NOT send the SAS voice prompt to vigor router B.
2.
Even no voice prompt, but the RTP traffic is still secured until the call ends.
Note:
If the incoming or outgoing calls do not match any entry on the phonebook, the
router will try to make the call "being protected". But, if the call ends up "unprotected"(e.g.
peer side does not support ZRTP+SRTP), the router will not play out a warning message.
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Phone Book
In this section, you can set your VoIP contacts in the “phonebook”. It can help you to make
calls quickly and easily by using “speed-dial”
Phone Number
. There are total 60 index entries
in the phonebook for you to store all your friends and family members’ SIP addresses.
Loop
through
and
Backup Phone Number
will be displayed if you are using Vigor2860 series for
setting the phone book.
Click any index number to display the dial plan setup page.
Available settings are explained as follows:
Item
Description
Enable
Click this to enable this entry.
Phone Number
The speed-dial number of this index. This can be any
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number you choose, using digits
0-9
and
*
.
Display Name
The Caller-ID that you want to be displayed on your
friend’s screen. This let your friend can easily know who’s
calling without memorizing lots of SIP URL Address.
SIP URL
Enter your friend’s SIP Address.
Dial Out Account
Choose one of the SIP accounts for this profile to dial out. It
is useful for both sides (caller and callee) that registered to
different SIP Registrar servers. If caller and callee do not
use the same SIP server, sometimes, the VoIP phone call
connection may not succeed. By using the specified dial out
account, the successful connection can be assured.
Loop through
Choose PSTN to enable loop through function.
Backup Phone Number
When the VoIP phone obstructs or the Internet breaks down
for some reasons, the backup phone will be dialed out to
replace the VoIP phone number. At this time, the phone call
will be changed from VoIP phone into PSTN call according
to the loop through direction chosen. Note that, during the
phone switch, the blare of phone will appear for a short
time. And when the VoIP phone is switched into the PSTN
phone, the telecom co. might charge you for the connection
fee. Please type in backup phone number for this VoIP
phone setting.
Secure Phone
ZRTP+SRTP
- It allows users to have encrypted RTP
stream with the peer side using the same protocol
(ZRTP+SRTP). Check this box to have secure call.
Cancel
Return to previous web page.
After finishing all the settings here, please click
OK
to save the configuration.
Note:
If the incoming or outgoing calls do not match any entry on the phonebook, the
router will try to make the call "being protected". But, if the call ends up "unprotected"(e.g.
peer side does not support ZRTP+SRTP), the router will not play out a warning message.
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Digit Map
For the convenience of user, this page allows users to edit prefix number for the SIP account
with adding number, stripping number or replacing number. It is used to help user have a
quick and easy way to dial out through VoIP interface.
Available settings are explained as follows:
Item
Description
Enable
Check this box to invoke this setting.
Match Prefix
It is used to match with the number you dialed and may be
modified by the action (add, strip or replace) with the
OP
Number
.
Mode
None
- No action.
Add
- When you choose this mode, the OP number will be
added before the match prefix number for calling out
through the specific route.
Strip
- When you choose this mode, the partial or whole
match prefix number will be deleted according to the OP
number. Take the above picture (Prefix Table Setup web
page) as an example, the OP number of
886
will be deleted
completely for the match prefix number is set with
886
.
Replace
- When you choose this mode, the OP number will
be replaced by the prefix number for calling out through the
specific VoIP interface. Take the above picture (Prefix
Table Setup web page) as an example, the prefix number of
03 will be replaced by 8863. For example: dial number of
“031111111” will be changed to “88631111111” and sent

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