Dialogic 4000 Series SU4.1 Manual de usuario descargar pdf (Pagina 49)

Dialogic® Diva® SIPcontrol™ Configuration

Default peer

for received

SIP calls

Enable this option if the selected peer type should be used as the

default peer. Calls from unconfigured SIP peers will be assigned to

this peer, and therefore are handled with these settings. This option

can also be changed via the Default Peer option on the main

SIPcontrol page.

Note: When a peer is selected as the default, the previously selected

default peer is automatically unselected.

Display name

to:

Enter the name to be sent in the "To" header of the INVITE message

on calls from the PSTN to SIP.

Display name

from:

Enter the name that is to be sent in the "From" header of the INVITE

message on calls from the PSTN to SIP. To send the calling party

number include an asterisk (*) in the display name. For instance, if

the display name is "Dialogic *" and the calling number is 123, then

the remote side receives "Dialogic 123". To include an asterisk in the

display name, enter "\*". To include a backslash enter "\\".

User name to:

You can enter a user name in front of the host name, e.g.,

thomas@dialogic.com. The user name is needed for the default route

when no called party number is transmitted, e.g., for Diva Analog

Media Boards.

If a call from SIP does not contain a user name, the name entered

here is transmitted to the receiver as the calling party number. This

applies to all references to PSTN in this section. (The opposite side can

either be PSTN or SIP.)

User name

from:

Enter the user name that is added to the SIP address when a number

from the PSTN is suppressed. You can also enter the complete SIP

address consisting of <username>@<local-IP/hostname>.

If a call from SIP does not contain a user name, the name entered

here is transmitted to the PSTN as the called party number.

Gateway

prefix:

You can configure this parameter only if you selected e-phone as

Peer type in the Edit SIP Peer Configuration window.

This prefix is added at the beginning of the address in the "Reply-To"

and "Contact" headers, which are copies of the "From" address. If this

string is not empty, the parameter "phone-context" will be added in

both headers.

Reply-To

expression:

You can configure this parameter only if you selected e-phone as

Peer type in the Edit SIP Peer Configuration window.

Enter the expression that may be necessary for the e-phone server to

handle the call. Normally, this is necessary to omit the 0 (zero) for

external calls and to manipulate the address so the e-phone server is

able to call back.

1 2 ... 44 45 46 47 48 49 50 51 52 53 54 ... 225 226

Comentarios a estos manuales

Sin comentarios

Dialogic 4000 Series SU4.1 Manual de usuario Pagina 49

Comentarios a estos manuales

Relacionado con productos y manuales para Gateways / Controladores Dialogic 4000 Series SU4.1