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

Key benefits
  • Scalable platform 1U
  • IP-PBX for 2 000 subscribers with VAS support
  • High-quality voice processing
  • Carrier class reliability
  • Up to 768 VoIP channels
  • Up to 16 Е1 flows (RJ-48)
  • Support of 2 built-in SD 8Gb
  • Hardware redudancy

Get price.Free equipment test
Documents and files
Physical parameters


SMG-1016M is used as a trunk gateway for interfacing of signal and media streams of TDM and VoIP networks. The gateway also might be used as an IP PBX with VAS support and a universal solution for infocommunication new generation networks (NGN). The wide function-set, strict compliance with requirements and standards and carrier class reliability allow service providers to solve most part of their objectives using SMG-1016M.

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.

Carrier class reliability
SMG-1016M provides high level of fault tolerance, uniform load distribution among submodules, power modules redundancy
and usage of up-to-date technologies based on parallel computing. The gateway will switch to a backup submodule in case of a primary submodule fault.

Functional compatibility
Strict compliance with up-to-date protocols’ requirements, recommendations and standards provides functional compatibility with a variety of equipment: digital PBX, IP PBX, Softswitches, VoIP gateways, SIP phones, programmable SIP clients, etc.

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

RADIUS routing
Intellectual call routing based on the billing system responses according to RADIUS protocol will help build flexible rules for calls 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 subnetworks, 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.

IP PBX with VAS support
Additional options for SMG-1016M gateway allows using it as a full-featured IP PBX with up to 2000 SIP subscribers connection and support for a wide range of value added services. A programmable IP PBX module ECSS-10 is dedicated to fast deployment of a VoIP node with a minimum of capital expenses. ECSS-10 and SMG-1016M might be used as a PBX of any level.


Calls management
  • Interaction with STUN-server on the SIP interface
  • Routing based on called number (CdPN) or calling number (CgPN)
  • 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 connection of trunk groups
  • Prefix for several trunk groups
  • Interactive voice menu (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 (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)

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 (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

  • INBAND, RFC 2833, SIP INFO, SIP NOTIFY transmission methods

  • Billing data is recorded in CDR file. CDR files are kept on a local HDD and remote FTP server
  • RADIUS Accounting
  • Supported billing systems: Hydra Billing, LANBilling, PortaBilling, NetUP, BGBilling (there is an opportunity of integration with other systems)

  • 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 LE*
  • V5.2 AN*

VoIP protocols
  • H.323*
  • H.248*

Capacity and performance
  • Up to 768 VoIP channels
  • Up to 16 Е1 streams (RJ-48)
  • Maximum load intensity 14 cps

  • 2 x 1000Base-X ports (2 slots for SFP modules)
  • 3 x 10/100/1000Base-T (RJ-45) ports
  • E1 (2 x CENTRONICS-36 connectors)
  • 2 SATA ports (for SSD modules installation)

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
  • Tracings are stored on HDD and USB storages
  • Emergency notification through SNMP

  • Black and white IP addresses lists
  • Attempts of access to device are logged
  • Automatic blocking by IP address after unsuccessful login attempts or/and access via http/https/telnet/ssh
  • List of permitted IP addresses for access to control of the device
  • Access rights delimitation – admin/user
  • Delimitation of rights to access calls records
  • Control of opposite RTP stream’s source IP address
  • Authentication of subscribers on RADIUS server and SIP registar
  • Digest authentication (RFC 5090, Draft-Sterman)
  • Digest authentication in RADIUS (RFC 5090, Draft-Sterman)

  • Operation in warm redundancy mode 1+1
  • The system switches the redundant part on automatically
  •  Automatic synchronization of main redundant module settings

Advanced SIP/SIP-T/SIP-I functionality
  • Registration and authentication of up to 3000 SIP subscribers 1
  • VAS support for up to 3000 SIP subscribers 1
  • 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

Value added services*
  • Call Forwarding
  • Call forwarding out of service (CFOS)
  • Call forwarding on no reply (CFNR)
  • Call forwarding unconditional (CFU)
  • Call forwarding on busy (CFB)
  • Call Transfer
  • Music on Hold (MOH)
  • Call Hold
  • Call Hunt
  • Call Pickup
  • Busy Lamp Field
  • Conference add-on (CONF)
  • Conference for a list of subscribers
  • 3-Way conference
  • Intercom
  • Paging
  • Outgoing calls restrictions
  • Egress communication by password (RBP)
  • Password activation (PWD ACT)
  • Password reset (PWD)

 Current firmware version 3.14.0

Documents and files


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.

Physical parameters

Range of operating temperatures
  • from 0 to +40°С
Relative humidity
  • 80% max
Noise level
  • from 44 to 60 dB
Power supply
  • AC:
  • DC:
  • 1 AC/DC power supply
  • 2 hot-swappable AC/DC power supplies
Power modules
  • AC, power module PM160-220/12 160W
  • DC, power module PM100-48/12 100W
Max. power consumption
  • 50W
Dimensions (WхHхD)
  • 430 х 45 х 260 mm
  • 19", 1U
Nett weight