Ru En
Designer and manufacturer of communication equipment
Russia (GMT +7)

VoIP Trunk Gateway SMG-1016M with IP-PBX support

Mass production
  • Scalable 1U platform
  • IP PBX for up to 2,000 subscribers with VAS support
  • High quality voice processing
  • Carrier-grade reliability
  • Up to 768 VoIP channels
  • Up to 16 streams Е1
  • Support for up to 2 embedded 8 GB SSDs
The SMG-1016M platform can be used as a trunk gateway for connecting signal and media flows of TDM and VoIP networks, IP PBX with support for VAS functions, and can also act as a universal solution for building new generation infocommunication networks. The wide function set, strict compliance with requirements and standards, as well as carrier-grade reliability allow service providers and carriers to solve most part of their objectives on the basis of SMG-1016M.

Scalability
SMG-1016M is a beneficial investment in the future of your project due to its scalability. The gateway supports up to 16 E1 streams (SS7, PRI, V5.2) and up to 768 VoIP channels.

IP PBX with VAS support
Additional options for SMG-1016М gateway allow using it as a full-featured IP PBX for up to 2,000 SIP subscribers with support for a wide range of value added services. A programmable IP PBX module ECSS-10 is designed for fast deployment of a VoIP node with a minimum of capital expenses (CAPEX). ECSS-10 and SMG-1016М might be used as a PBX of any level.

Carrier-grade reliability
Uniform load distribution between submodules, redundant power supplies, as well as the use of modern technologies based on parallel computing provide a high level of fault tolerance of the SMG-1016M trunk gateway with automatic switching to a backup submodule in the event of any system submodule failure or the power source.

Functional compatibility
The strict compliance with requirements of up-to-date protocols, recommendations and standards provides functional compatibility of SMG-1016M with a variety of equipment: digital PBX, IP PBX, Softswitch, VoIP gateways, SIP phones, software SIP clients, etc.

Media streams transcoding
The hardware transcoding helps to negotiate media streams with different VoIP codecs which are used in up-to-date networks.

RADIUS routing
Intellectual call routing based on billing system responses via the RADIUS protocol allows creating flexible methods of call processing.

Intellectual protection of IP networks
The intellectual protection against unauthorized external SIP subscribers connection and connections via http/https/telnet/ssh is realized on the SMG-1016M (Dynamic Firewall, Static Firewall, black and white lists of IP addresses and subnets, etc.). For additional defense, SMG-1016M is compatible with session border controllers (e.g. SBC-1000) that are used as a firewall for VoIP networks.
Specifications
Documents and files
Warranty
Calls management
  • Interaction with STUN-server on the SIP interface
  • Routing based on called number (CdPN) and/or calling number (CgPN)
  • Routing by the access category
  • Number modifications before and after routing
  • Call recording according to number mask and dialplan1
  • Use of multiple dialplans
  • Subscriber lines restriction
  • Subscriber service mode settings
  • Trunk group cut-off
  • Call management via RADIUS1
  • Direct forwarding for trunk groups
  • Prefix for several trunk groups
  • Interactive Voice Response (IVR)1
  • Uploading/downloading of configuration as a single file
  • Lines limiting for SIP interface
  • Egress and ingress lines restrictions for a subscriber
  • Ingress load limiting CPS (calls per seconds) for a trunk group
Voice codecs
  • G.711 (a-law, µ-law), G.729 (A/B), G.723.1, G.726 (32 Kbps)
Video processing
  • Video stream transmitting in the Video Offroad mode
Fax transmission
  • T.38 Real-Time Fax, G.711 (a-law, µ-law) pass-through
Voice standards
  • VAD (Voice Activity Detection)
  • CNG (Comfort Noise Generation)
  • AEC (Acoustic Echo Cancellation, G.168 recommendation)
  • AGC (Automatic Gain Control)
Quality of service (QoS)
  • Diffserv and 802.1p priorities assignment for SIP and RTP
  • Dynamic and static jitter buffer
  • Ingress/egress traffic rate limiting
DTMF
  • INBAND, RFC 2833, SIP INFO, SIP NOTIFY transmission methods
Billing
  • Billing data is recorded in CDR file. Simultaneously, CDR file is recorded to a local HDD and remote FTP server
  • RADIUS Accounting
  • Supported billing systems: Hydra Billing, LANBilling, PortaBilling, NetUP, BGBilling (possible integration with other systems)
Flexibility
  • Multiple network interfaces creation for telephony (SIP, RTP) with different IP addresses
  • Operation with multiple numbering plans
  • Signal SS7 channel redundancy
  • Voice activity control (by the presence of RTP or RTCP)
  • Individual routing for streams of a single SS7 linkset
TDM protocols
  • SS7
  • PRI (Q.931)
  • Q.699 (PRI and SS7 interaction)
  • V5.2 LE1
  • V5.2 AN1
VoIP protocols
  • SIP, SIP-T/SIP-I, SIP-Q
  • H.3231
  • SIGTRAN (M2UA, IUA)2
  • H.2482
Capacity
  • Up to 768 VoIP channels
  • Up to 16 E1 streams (CENTRONICS-36)
  • Maximum load intensity 14 cps
Interfaces
  • 2 × 1000BASE-X ports (2 slots for SFP modules)
  • 3 × 10/100/1000BASE-T ports (RJ-45)
  • E1 (2 × CENTRONICS-36 connectors)
  • 1 USB 2.0 port
  • 1 console port (RS-232)
  • 2 × SATA ports (for SSD storage modules)
Phone book
  • Retrieving Display Name from LDAP server
Management and monitoring
  • E1 and VoIP channels monitoring in web interface
  • Management of channels and SS7 links in web interface
  • Alarm logging with the opportunity to save entries to syslog server
  • Storing traces on SSD and USB drives
  • Emergency notification through SNMP
  • Automatically enable logging after the gateway restart
  • Monitoring of web interface active user sessions
Security
  • Black and white IP addresses lists
  • Logging of all access attempts to the device
  • Automatic blocking by IP address after unsuccessful login attempts and/or access via http/https/telnet/ssh
  • List of permitted IP addresses for access to control the device
  • Access rights delimitation – admin/user
  • Delimitation of access rights to calls records
  • Control of opposite RTP stream source IP address
  • Digest authentication (RFC 5090, Draft-Sterman)
  • Digest authentication in RADIUS (RFC 5090, Draft-Sterman)
Value added services1
  • Call Forwarding
    • Call Forwarding Out of Service (CFOS)
    • Call Forwarding on No Reply (CFNR)
    • Call Forwarding Unconditional (CFU)
    • Сall Forwarding on Busy (CFB)
    • Forwarding by day of week and time of day
  • Call Transfer
  • Music on Hold (MOH)
  • Call Hold
  • SIP-forking support for SIP subscribers
  • Voice Notification
  • Call Hunt
  • Call Pickup
  • Call Parking
  • Busy Lamp Field
  • Conference add-on (CONF)
  • Conference for a list of subscribers
  • 3-Way conference
  • Intercom
  • Paging
  • Outgoing calls restrictions (Out Calls Restrict)
  • Egress communication by password (RBP)
  • Password activation (PWD ACT)
  • Password reset (PWD)
  • Voice mail
  • One Touch Record
  • Do Not Distrurb (DND)
  • Blacklist
  • Anonymous call
  • Reject anonymous calls
  • Reminder
Advanced SIP/SIP-T/SIP-I functionality
  • Registration and authentication of 2,000 SIP subscribers1
  • VAS support for 1,000 SIP subscribers1
  • SIP and SIP-T/SIP-I interaction
  • Trunking and subscriber registration of SIP trunks
  • Transit registration of subscribers on SIP trunk with switching to local service mode in case of server unavailability
Physical specifications and environmental parameters
  • Operating temperature range - from 0 to +40 °C
  • Relative humidity - up to 80 %
  • Noise level - from 44 to 60 dB
  • Supply voltage
    • DC: 36–72 V
    • AC: 100–240 V, 47–63 Hz
  • Power options:
    • 1 AC/DC power supply;
    • 2 hot-swappable AC/DC power supplies.
  • Power modules
    • DC, power module PM100-48/12 100 W
    • AC, power module PM160-220/12 160 W
  • Power consumption no more than 50 W
  • Dimensions (W × H × D) - 430 × 45 × 260 mm
  • Form factor - 19", 1U
  • Weight - 3.2 kg

1Оptional
2 Not supported in the current firmware version
Operational lifetime of the ELTEX equipment
In development
1
Pre-production
2
Mass production
3
Mass production is over
4
Sold out
5
Support is over
6
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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