The multi-site voip domain – AltiGen MAXCS 7.0 Update 1 ACM Administration User Manual
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The Multi-site VoIP Domain
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MaxCS 7.5 Administration Manual
The Multi-site VoIP Domain
Note:
This feature is not applicable to the MaxCS Private Cloud service.
A group of AltiGen systems can form a VoIP domain where they share the same global extension directory and
call routing rules. The VoIP domain is based on VoIP framework and uses IP tie-trunks to interconnect among
different sites.
A domain is created in MaxAdmin. Here, a system is designated as the domain Master. Other AltiGen systems
can then be added to a VoIP domain.
The VoIP domain Master maintains global configurations and propagates the configurations to all the members
belonging to this domain automatically. Any changes in the global configuration are propagated in real time to
the other members in the VoIP domain.
Note:
A multi-site installation requires an AltiGen Enterprise license.
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 the destination supports SIP 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.
Parameter
Description