Dialogic® 4000 Media Gateway Series Reference Guide
page 23
Enhanced
You may configure the parameters shown in the graphic and explained below:
Default SIP to
PSTN peer:
Enable this option if the selected peer type should be used as default peer. Calls from unconfigured
SIP peers will be assigned to this peer, and therefore are handled with these settings. If several
peers are configured as default, the Diva SIPcontrol software takes the first to transmit the call.
Display name to:
Enter the name that is 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 may enter a user name in front of the host name, e.g., [email protected]. The user name
is needed for the default route when no called party number is transmitted, e.g., for Dialogic
®
Diva
®
Analog Media Boards.
If a call from SIP does not contain a user name, the name entered here is transmitted as calling
party number to the PSTN.
User name from:
Enter the user name that is added to the SIP address when a number from the PSTN is suppressed.
You may 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 as called
party number to the PSTN.
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.
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