Multi-Site Voip Management - Enterprise Manager - Altigen ACM 6.7 Administrator's Manual

Overview
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 providers 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 feature helps to resolve DTMF tone detection and regeneration when
using G.723.1 or G.729 codec. Basically 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 configured as Global extension, the VoIP domain controller will propagate
this extension to all member systems. 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 administrator 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.
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MAXCS ACM 6.7 Administration Manual