Altigen MAX Communication Server ACC 6.5, Administration Manual

Page: 356 / 488

Chapter 25: Enterprise VoIP Network Management
342
MAXCS ACC 6.5 Administration Manual
Define the attributes for the entry:
Parameter Description
Server ID A unique dialing number to connect to the remote
server. The server could be AltiServ, a 3rd-party
VoIP gateway, or an AltiGen-certified 3rd-party VoIP
device.
Server Name A descriptive name of up to 15 characters to identify
the server. This name may be used by Caller ID.
Server IP Address
a
The remote server’s address. If the server has
multiple IP addresses, enter the one that other
servers will use to communicate to this system.
This IP address format is recommended over DNS
names, since with the IP address, the application
does not need to resolve the name. DNS name is also
posted in this field.
Remote Ext. Length The length of extension digits at the remote location.
Valid entries are None - 7, with “None” meaning not
specified. Specifying the remote extension length is
optional but highly recommended, since this
information tells the system how long to wait for
another entry before sending the digits.
Dialing Scheme Overlapping (ATGN) allows the terminal to omit
part of the digits required to complete a call while
buffering the remaining digits. This results in faster
response time, but it only works if the other end is
also an AltiServ system.
Enbloc allows the system to buffer all of the digits
required to complete a call.
Protocol SIP Select if destination supports SIP protocol.
H323 Select if destination supports H.323 protocol.
SIP/TCP Select if adding an entry to the table to
support Microsoft Exchange integration.
Codec Select which codec profile to use. If the selected
profile is incompatible with the remote end, the call
will not go through.
If you create two items that point to the same IP
address, they must also use the same codec.
Specifying a different codec is an invalid
configuration. MAXCS will always use the codec
defined in the first item.
Hop Off Allowed Choosing Yes allows calls from this remote system
to hop off to the PSTN by using the trunks in this
system. Hop-off capability can be enabled or
disabled on a per IP Dialing Table Location basis.
SIP Source Port Used by UDP only. Choose the SIP source port.
SIP Destination Port Used by UDP only. Is 10060, by default.