ASUS AX-112 VoIP Phone Adapter

Specifications
Performance

Can handle up to 2 low bit rate VoIP call under normal LAN <---> WAN data traffic.
LAN--WAN bridge throughput : 5 Mbps

VoIP Features

Vocal codec:
G.711 u/A
G.723.1
G.729A
Max concurrent VoIP channel:
G.723.1: 1 Channel
G.729A: 2 Channel

Voice related capability:
AEC : Auto echo cancellation, G.168/16ms
VAD: Voice activity detection (silent suppression transmission)
CNG: Comfortable noise generation
Packet concealment

Jitter buffer:
Fix length jitter buffer: Length definable
Adaptive jitter buffer:

DTMF method:
RFC2833
SIP Info
In-band audio

G.3 Fax support: T.38 ,G.711 pass through
Caller ID:
DTMF/FSK generation
Calling tone generation: Dial/ Busy/ Ring back tone
FXS hook detection: Flash key detection for call transfer
Major call features: Call hold,Call transfer, Call forward, Call return, Do Not Disturb
3-way Conference (G.729/G.711)
8 entries of speed dial, 8 entries of phone book dialing. Allows user to define destination IP and port for each phone book entry
Volume control : Tx/Rx gain definable (-12db ~ +18db)

Network Management

WAN/LAN operation: Bridge mode
Support VPN pass through
PPPoE
QoS: ToS Tag, ToS tag. 802.1Q ,802.1p
Authentication : Text/MD5/Digest-MD5
NAT traversal : STUN, Outbound proxy
WAN IP:
DHCP client
Static IP assignment

Management

Web-based GUI
Firmware upgrade and configure by TFTP and HTTP
Auto provisioning capability (Customization option)
LEDs for SIP activity and status indication.
Basic configuration by IVR, can setup network , WAN settings : Dynamic IP Assignment , IP Address , Subnet Mask , Default Gateway , and report WAN settings by phone set

Reset button Press the button and hold for 5 second will restore factory default value

DOWNLOAD LINK:

http://sg.asus.com/Networks/Wired_Routers_Switches/AX112/#download