VoIP Gateway TAU-24.IP

Key features

Office PBX functionality
High-quality sound
Current and voltage port protection
Measurement of subscriber line physical parameters
The maximum length of line – 6 km

Multiport Access VoIP-gateways TAU series are designed and suitable for voice and facsimile information transmission via the IP-network. Gateways provide customers with high-quality Telephone Communications with Value Added Services Support: call-forward, call waiting, 3-way conference, pick up, group call, call line identification presentation etc.

High quality Voice
The high quality of sound is ensured by the use of the high- performance hardware, support for main audio codecs used in VoIP networks (G.711, G.723.1, G.726, G.729), echo cancellation function, use of silence detector, comfort noise generation, DTMF signals reception and generation and prioritization mechanisms (QoS).

Redundancy
In case of the loss of main SIP server connection, TAU switches to the redundant SIP server automatically with monitoring of service capability of the main one. If there is a connection loss with both servers, local switching among gateway subscribers is saved.

Usability
A friendly multilingual management interface and support for group management means based on TR-069 and DHCP (DHCP autoprovisioning) enable easy exploitation of unlimited number of TAU on an operator’s network.

Eltex.EMS management system
Eltex.EMS is a unified management system for monitoring and control of a large number of gateways on a network. The system provides centralized management of a gateway group and its ports monitoring via the unified web interface.

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VoIP protocols

SIP
SIP-T
H.323

Voice codecs

G.729 (A, B)
G.711 (a-law, µ-law)
G.723.1 (6.3/5.3 Kbps)
G.726 (32 Kbps)

Fax

T.38 UDP Real-Time Fax
G.711 (a-law, µ-law) pass-through

Voice standards

VAD (voice activity detector)
CNG (comfort noise generation)
AEC (echo cancellation in accordance with ITU-T G.168)
AGC (automatic gain control)
PLC (packet loss concealment)

Features

SIP server authentication with common username and password for all subscribers
SIP server authentication with individual username and password for each subscriber
Support for redundant SIP servers
Support for Outbound SIP servers from DHCP Option 120
Direct routing to the unregistered devices on a SIP server
Internal switching is saved in case of SIP server connection loss
Locally Supplementary Services’ processing (distributed mini PBX mode)
Regular expressions in Dialplan
Caller and called numbers modifications
Distinctive ring service
User tone signals
Limitation of simultaneous connections
CPC (Calling Party Control): disconnect signal by circuit disruption
Support for pay phone
Support for operation behind NAT (STUN, PublicIP)
Signal generation when a handset is off-hooked
Supplementary Services management via phone added codes
Applying of settings without reboot
Forming of DHCP Option 82, Agent client circuit ID, Agent remote ID suboptions

Quality of service (QoS)

4 priority queues
Packet distribution to queues based on 802.1p and/or DSCP
Assigning of DSCP and 802.1р priorities for SIP and RTP packets

Value Added Services

Caller line identity presentation (CLIP)
Issuing of a caller name and time of a call in FSK mode
Calling line identification restriction (CLIR)
Call Transfer
Call Pickup
3-Way Conference
Hotline/Warmline
Call Waiting
Call Forward (CFU, CFB, CFNR, CFOOS)
Call Group
Call Hold
Music on Hold (MOH)
Message-waiting indicator (MWI)
Do not Disturb (DND)
IMS (3GPP TS 24.623) for Call Hold, Call Waiting, 3-Way Conference, Hotline, Call Transfer

Network functions

802.1Q
Possibility to use different VLAN for signalling, RTP and management
SNTP
Local and external DNS
STP
LLDP
Dual homing redundancy
IPSec
Firewall

Types of connections

Static IP address
DHCP client
PPPoE client
РРТР client

Remote monitoring

HTTP/HTTPS
SNMP
TR-069

Configuring

HTTP/HTTPS, FTP/FTPS, TFTP
Auto update of the firmware and configuration (DHCP options 43, 66 and 67)
Command line interface (CLI) via Telnet, SSH, Console port RS-232
Parameters configuring via SNMP (Eltex.EMS management system)
Parameters configuring via TR-069

Diagnostics

Syslog
Subscriber lines parameters testing
Checking for a phone available on the line

Statistics

Detailed statistics per port
Call history

Security

Username and password control
Access rights differentiation: admin/user
Configuration file encryption
Access to WEB via RADIUS authentication
Access to WEB only via HTTPS

Interfaces

FXS ports - 24
FXO ports - up to 32
Type of FXS/FXO port connector - TELCO-50
Ethernet 10/100/1000BASE-T ports (RJ-45) - 2
Ethernet 1000BASE-X ports (SFP) - 1
RS-232 console port - 1
Type of RS-232 console port connector - RJ-45
WAN connections types - Static, DCHP, PPPoE/PPTP

Physical features and ambient conditions

Power supply - 220 V AC or 48/60 V DC
Power consumption at 0.2 Erl - ≤ 30 W
Power consumption at 1 Erl - ≤ 35 W
Dimensions (W×H×D), mm - 430×45×134
Operating temperature - from 0 to +40 °С
Weight - 3 kg
Form factor - 19", 1U
Operating humidity - up to 80 %

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Descriptions

Datasheet

Software

Firmware version 2.21.0

Reserve firmware

Firmware version 2.20.9

Firmware version 2.20.7

Firmware version 2.18.0

Firmware version 2.16.0

Firmware version 2.13.1

Documentations

Operation manual version 2.21.0

Downloads

Operational lifetime of the ELTEX equipment

Regardless of the operational lifetime stage, Eltex provides a 12 months warranty on all its telecommunication equipment.
During the warranty period the manufacturer ensures technical support and free-of-charge repair at the Enterprise which is situated in Novosibirsk.
As part of the warranty service, technical support is provided on the first-in first-out principle.
The priority support packages of 8/5 and 27/7 types are subjects to additional charges.

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