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Page 371 / 794 Scroll up to view Page 366 - 370
Vigor2860 Series User’s Guide
357
After finishing all the settings here, please click
OK
to save the configuration.
3.15.2 SIP Accounts
In this section, you set up your own SIP settings. When you apply for an account, your SIP
service provider will give you an
Account Name
or user name,
SIP Registrar, Proxy,
and
Domain name
. (The last three might be the same in some case). Then you can tell your folks
your SIP Address as in
Account Name@ Domain name
As Vigor VoIP Router is turned on, it will first register with Registrar using
AuthorizationUser@Domain/Realm. After that, your call will be bypassed by SIP Proxy to the
destination using AccountName@Domain/Realm as identity.
Note:
Selection items for
Ring Port
will differ according to the router you have.
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Available settings are explained as follows:
Item
Description
Index
Click this link to access into next page for setting SIP
account.
Profile
Display the profile name of the account.
Domain/Realm
Display the domain name or IP address of the SIP registrar
server.
Proxy
Display the domain name or IP address of the SIP proxy
server.
Account Name
Display the account name of SIP address before @.
Codec
Display the codec type for the account.
Ring Port
Specify which port will ring when receiving a phone call.
Status
Show the status for the corresponding SIP account.
R
means such account is registered on SIP server
successfully.
–
means the account is failed to register on
SIP server.
STUN Server
Type in the IP address or domain of the STUN server.
External IP
Type in the gateway IP address.
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359
SIP PING interval
The default value is 150 (sec). It is useful for a Nortel
server NAT Traversal Support.
Click any index link to access into the following page for configuring SIP account.
Available settings are explained as follows:
Item
Description
Profile Name
Assign a name for this profile for identifying. You can type
similar name with the domain. For example, if the domain
name is
draytel.org
, then you might set
draytel-1
in this
field.
Register via
If you want to make VoIP call without register personal
information, please choose
None
and check the box to
achieve the goal.
Some SIP server allows user to use VoIP
function without registering. For such server, please check
the box of
Call without Registration
. Choosing
Auto
is
recommended. The system will select a proper way for your
VoIP call.
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SIP Port
Set the port number for sending/receiving SIP message for
building a session. The default value is
5060.
Your peer
must set the same value in his/her Registrar.
Domain/Realm
Set the domain name or IP address of the SIP Registrar
server.
Proxy
Set domain name or IP address of SIP proxy server. By the
time you can type
:port
number
after the domain name to
specify that port as the destination of data transmission
(e.g.,
nat.draytel.org:5065
)
Act as Outbound Proxy
Check this box to make the proxy acting as outbound proxy.
Display Name
The caller-ID that you want to be displayed on your friend’s
screen.
Account Number/Name
Enter your account name of SIP Address, e.g. every text
before @.
Authentication ID
Check the box to invoke this function and enter the name or
number used for SIP Authorization with SIP Registrar. If
this setting value is the same as Account Name, it is not
necessary for you to check the box and set any value in this
field.
Password
The password provided to you when you registered with a
SIP service.
Expiry Time
The time duration that your SIP Registrar server keeps your
registration record. Before the time expires, the router will
send another register request to SIP Registrar again.
NAT Traversal Support
If the router (e.g.
,
broadband router) you use connects to
internet by other device, you have to set this function for
your necessity.
None –
Disable this function.
Stun
– Choose this option if there is Stun server provided
for your router.
Manual
– Choose this option if you want to specify an
external IP address as the NAT transversal support.
Nortel
– If the soft-switch that you use supports Nortel
solution, you can choose this option.
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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.
Ring Port
Set Phone 1 and/or Phone 2 as the default ring port(s) for
this SIP account.
Ring Pattern
Choose a ring tone type for the VoIP phone call.
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.

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