VoIP Trunk Gateway SMG-3016 with IP-PBX support

Key benefits
  • Scalable platform 1U
  • IP PBX for 3 000 subscribers with VAS support
  • High-quality voice processing
  • Carrier class reliability
  • Up to 768 VoIP channels
  • Up to 16 Е1 streams (RJ-48)
  • Support for 2 HDD SATA 2.5
  • Hardware redundancy

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Physical parameters


Hybrid platform SMG-3016 is used as a trunk gateway for reciprocation of signal and media flows of TDM and VoIP networks. The gateway also might be used as an IP PBX with value added services (VAS) support and a multipurpose solution for info communication new generation networks (NGN). The wide function-set, strict compliance with require-ments and standards and carrier class reliability allow service providers and carriers to solve most part of their objectives on the basis of SMG-3016.

 SMG-3016 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-3016 gateway allow using it as a full-featured IP PBX for up to 3000 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. ECSS-10 and SMG-3016 might be used as a PBX of any level.

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

Carrier class reliability
 SMG-3016 provides high level of fault tolerance, uniform load distribution among submodules, usage of up-to-date technologies based on parallel computing and power modules redundancy. In case of a primary submodule fault, the gateway switches to a backup submodule.

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

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-3016 (Dynamic Firewall, Static Firewall, black and white lists of IP addresses and subnetworks, etc.). For additional defense, SMG-3016 may be used in coupling with with session border controller (e.g. SBC-1000) that is used as a firewall for VoIP networks.

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


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 dialplan*
  • Use of multiple dialplans
  • Subscriber lines restriction
  • Subscriber service mode settings
  • Trunk group cut-off
  • Call management via RADIUS*
  • Direct forwarding for trunk groups
  • Prefix for several trunk groups
  • Interactive voice response (IVR)*
  • 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

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

  • Billing data is recorded as a CDR file that is kept on a local HDD and remote FTP server simultaneously
  • 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 dialplans
  • 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.3231
  • H.2482

Capacity and performance
  • Up to 768 VoIP channels
  • Up to 16 E1 streams (RJ-48)
  • Maximum load intensity — 120 cps
  • RAM2GB

  • 16 E1 ports (RJ-48)
  • 2 ports of 10/100/1000 Base-T (RJ-45) / 1000 Base-X (SFP)
  • 2 ports of 10/100/1000 Base-T (RJ-45)
  • 2 USB 2.0 ports
  • 2 slots for SATA HDD 2,5'’

Management and monitoring
  • E1 and VoIP channels monitoring via web interface
  • Channels and SS7 links management via 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
  • Console port RS-232 (RJ-45)
  • Allocated management port (OOB) 10/100/1000BASE-T (RJ-45)

  • Black and white IP addresses lists
  • Attempts of accessing the device are logged
  • Automatic blocking by an IP address after unsuccessful login or/and access attempts via http/https/telnet/ssh
  • List of permitted IP addresses for access to control the device
  • Access rights delimitation – admin/user
  • Delimitation of rights to access calls records
  • Control for 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)

Advanced SIP/SIP-T/SIP-I functionality
  • Registration and authentication of up to 3000 SIP subscribers*
  • VAS support for up to 3000 SIP subscribers*
  • SIP and SIP-T/SIP-I interaction
  • Trunking and subscriber registration of SIP trunks
  • Transit registration of subscribers on SIP trunk with switching to a local servicing in case of server unavailability

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

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)
  • Do not disturb
  • Blacklist

**Not supported in the current firmware version — 3.15.0

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: 220V+-20%, 50 Hz
  • DC: -48V+30%-20%
  • 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 340 mm
  • 19", 1U
Nett weight
5,3 kg