Altigen ACM 6.7, Руководство администратора

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Defining the IP Dialing Table
330 MAXCS ACM 6.7 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 MAXCS, 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 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 a MAXCS 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.
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.
Publish as a global
entry If you are adding a system or 3rd-party VoIP device that is not
part of the VoIP domain, but you want it to be seen by all
servers in the domain, check this box. (The entry will appear as
“Global” in the Type column.) You can also globalize it later by
selecting the entry in the IP Dialing Table and clicking the
Publish as Global button below the table.