AltiGen MAXCS 7.0 Update 1 ACM Administration - Manual (Page 38)

Chapter 1: New Features
22 MaxCS 7.5 Administration Manual
H.323 Tie-Trunk Support - Ensures backward compatibility to systems using AltiGen’s AltiWare ver-
sions prior to 5.1.
IP Extension Auto Failover - when an IP extension is unreachable, the system will automatically fail
over to a pre-configured Mobile Extension.
IP Group Paging - allows the use of voice paging to IP phone users in a group.
NAT Configuration for SIP/H.323 - When MAXCS is behind NAT with a private IP address, this feature
helps to resolve IP address resolution problems when communicating with an external VoIP device.
Silence Detection and Suppression - when silence suppression is enabled and silence is detected, the
system stops sending packets to the other side. The other side does not receive any packets and plays
silence.
VoIP Hop-Off Call Support - allows an extension to access a PSTN trunk on the remote system and
“hop off” to dial an outside telephone number. This hop off feature can be enabled or disabled on the
remote system. Outcall restrictions for hop off calls are configurable.
SIP Trunk Support - MAXCS enables AltiGen’s system to connect to IP-based trunking service provid-
ers via SIP.
SIP NAT Traversal - Allows MAXCS to connect to a remote SIP phone or IPTalk behind NAT without
changing the NAT setting at the remote location.
Support for RFC 2833 (DTMF payload embedded with RTP) - Supported in SIP trunks only. This fea-
ture helps to resolve DTMF tone detection and regeneration when using G.723.1 or G.729 codec. Basi-
cally low bit rate compression will distort DTMF tone during compression. The far end device may not be
able to recognize the DTMF digits. RFC 2833 specifies a separate RTP payload format to carry DTMF
information to ensure the other side can recognize the tone properly.
Support for both SIP and H.323 Tie Trunk - When setting up a system-to-system VoIP tie trunk, either
SIP or H.323 protocol can be used.
Multi-Site VoIP Management - Enterprise Manager
Multi-site management through Enterprise Manager includes:
VoIP domain - when networking multiple AltiGen systems from different sites, one system can be
assigned as VoIP domain controller to propagate configuration data to member systems.
Directory Synchronization - when a new extension is added to one of the member systems and config-
ured as Global extension, the VoIP domain controller will propagate this extension to all member sys-
tems. Every member system within the VoIP domain will be able to see the extension number plan of
other systems.
Multi-site Call Routing - when a user dials an extension number that is not a local extension number,
the system will search the Domain extension list. If a list is found, the system will dial the number by
using the IP address and extension number stored in the Domain extension list.
Domain User Management - The VoIP domain controller can resolve the conflict if duplicated extension
numbers are created in different member systems. This feature also manages extension relocation.
When an extension user is relocated to another member system, its voice mail and greeting can be
moved along with it.
Global Least Cost Routing - when multiple systems are in different area codes or countries, the admin-
istrator can set up Global Least Cost Routing to route long distance or international calls through member
systems. The routing rules are propagated to all members automatically.
Global Dial-by-Name and Greeting Synchronization - Caller using the dial-by-name feature from any
system within the VoIP domain can search the entire global directory. The global extension’s greeting is
replicated to all systems within the VoIP domain.
Global Extension Relocation by User - When a global extension user travels to any site, the user can
dial #27 to log in to the local server. Enterprise relocates the user’s extension setting and voice mail to
the local server and activates the extension as a physical extension. All member systems receive an