Bedienungsanleitung ZyXEL Gateway 400 (Seite 28 von 68) (Englisch)

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Gateway 400 User Guide | Configuration

Select provider: Select the desired VoIP provider (Internet telephony provider) from one of the pre-

configured VoIP providers listed here. New providers can be set up under VoIP Pro-

vider. See "VoIP Provider" on page 27.

Choose Account Type: Select the type of account provided by your VoIP provider.

Display name: Enter a user-defined name.

Access name: Enter the data provided by your VoIP provider.

Authentication name: Enter the data provided by your VoIP provider.

Password: Enter the data provided by your VoIP provider.

Registration time: The default Registration time of “900” seconds can remain unchanged.

Area Code (optional): Enter the country code (“49” for Germany) in the first field and the area code

in the second field. If the country code is used, the area code must be entered with-

out a leading “0”; if the country code is not used, the area code must be entered

with a leading “0”.

Call number: Fixed network number without area code, as provided by your VoIP provider.

Not all VoIP providers use this setting. If no connection can be initialized with the

specified data, you may have to enter the fixed network number with the area code

(area code with a leading “0”) here. In this case, the fields under Area Code would

remain empty.

PABX extension: Only relevant for the account type SIP Trunk. Here you enter the call number range

for the (PABX) extensions, e.g., “11-99”

Default extension: Only relevant for the account type SIP Trunk. Here you enter the DID number for

the intercept position (central attendant), e.g., “0”.

Activate: Select this check box if you want to use this call number.

Use Registrar: Select this check box if you want to use the registrar server of the VoIP provider.

Some VoIP providers do not require a registrar; in such cases, this function must be

disabled.

Enter the required VoIP account data and click Save. Registering the account may take some time to

complete. Please wait until the previous screen dialog Internet telephony accounts is displayed

again.

5.5.4 VoIP Services

The Session Initiation Protocol (SIP) is a network protocol for setting up, controlling and clearing

down a communication session between two or more subscribers. From the drop-down list Service

Implementation, select the mapping that corresponds to the format used by your VoIP provider.

Description of the header fields:

INVITE: INVITE is used for making connection requests to the server; it includes the SIP ad-

dress of the recipient

From: SIP address of the sender

To: SIP address of the recipient

P-Preferred-Identity: The “Identity Header” is used to define which Public User Identity (call num-

ber) is used for the connection setup.

P-Asserted-Identity: The usage of P-Asserted-Identity and P-Preferred-Identity depends on the VoIP

provider; the value itself can be set via the mapping. For SIP trunks, the call number

can be “call number + default extension” or “call number + DID number”.

Privacy: Setting for caller ID display at the remote end:

“None”: caller ID restriction disabled, i.e., CLIP (Calling Line Identification Presenta-

tion)

“ID”: caller ID restriction enabled, i.e., CLIR (Calling Line Identification Restriction)

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