AltiGen comm ACM 5.1 - Manual (Page 31)

Chapter 1: Overview
AltiWare ACM 5.1 Administration Manual 17
Monitor List - lets you configure an extension’s privilege to see other extension’s call
activity through AltiView or AltiAgent.
Password Security - allows administrators to lock extensions that have been
“attacked” with false password attempts and to set default system passwords for newly
created or newly assigned extensions.
Out Call Routing Configuration - allows outgoing calls to be directed to particular
trunk routes, based on parameters assigned in the Out Call Routing table.
Remote Administration - a version of the AltiWare Administrator application that can
be installed on a Windows 2000/2003/XP client computer to remotely administer one or
more systems.
Transmit Extension Calling ID - each extension can be configured with a calling ID.
When an outgoing call is made by this extension through PRI or IP trunks, the calling ID
is displayed as the Caller ID to the receiving caller.
Voice over IP Features
VoIP features include:
Bandwidth Control for VoIP Sessions - Each server can configure the maximum
concurrent VoIP sessions based on its Internet or intranet bandwidth. This feature is to
ensure that voice quality will not be impacted if too many VoIP sessions are connected
at the same time.
Codec Profile - Multiple codec profiles with different settings can be created and applied
to different locations. Each profile can have a different codec, jitter buffer, and packet
length to accommodate different IP connections.
DNIS Name Display and Routing over IP Tie Trunk - allows for DNIS information to
be transferred to the other system when routed over IP tie-trunks. DNIS name of
matched entry can be displayed at AltiConsole, AltiView, AltiAgent, and handset.
Caller ID/Name Sent Over IP Tie Trunk - SIP supports sending the caller’s name, so
SIP and H.323 calls may display different caller ID information.
DTMF payload embedded with RTP (RFC 2833) - this feature helps to resolve DTMF
tone detection and regeneration when using G.723.1 or G.729 codecs. Low bit rate
compression can distort DTMF tones during compression and cause the far end device to
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.
Dynamic Jitter Buffer - due to various delays in the IP network, audio packet streams
may be delivered late or out of order. The system is able to buffer incoming packets and
re-sequence them by maintaining a queue. This queue is adjusted dynamically to
accommodate different network environment characteristics.
Echo Cancellation - due to bandwidth limitations and device loading, long delays may
occur during packet delivery process, which worsens the echo effect voice speech. Echo
cancellation is provided to maintain reasonable voice quality.
G.711 Codec - toll quality (64K) digital voice encoding, which guarantees
interoperability and better voice quality.
G.723.1 Codec - a dual rate audio encoding standard, which provides near toll quality
performance under clean channel conditions.
G.729 A+B Codec - speech data encoding/decoding standard of 8 Kbps.