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

Enterprise IP PBX SMG-500

Mass production
Key Benefits
  • IP PBX for up to 500 subscribers
  • Up to 100 calls simultaneously
  • Up to 4 E1 flows (RJ-48)
  • 4 LAN ports
  • Call center functionality
  • Call recording
SMG-500 is an enterprise PBX for 500 subscribers with a full set of Supplementary Services (SS).

The basic configuration of the enterprise IP PBX SMG-500 is designed to connect up to 250 SIP subscribers and can be extended to connect up to 500 subscribers when purchasing the appropriate software1. The E1 ports and SIP trunks can be used for connection to PSTN. Analog phones are connected to SMG-500 via subscriber VoIP gateways, IP phones – directly via the company network. Call recordings and CDR files are stored on the SD card or USB drive. It is also possible to automatically upload files to external data storage or an FTP server.

Networking of separated offices
The SMG-500 allows clients to organize an enterprise telephone network between remote offices of the company with minimal costs. Landline phone numbers in all offices remain the same, as customers will continue calling the known numbers. Employees from different offices can call each other on short numbers absolutely free, thus reducing the cost of intercity and international calls.

Multiservice platform
The variety of services allows clients to create the most efficient individual call processing scenarios. The SMG-500 supports conference calls, call recording, multiple channels and interactive voice menu.

Functional compatibility
Strict compliance with modern protocols, recommendations and standards ensures 100% functional compatibility of SMG-500 with equipment from different vendors: digital PBX, IP PBX, Softswitch, VoIP gateways, SIP phones, SIP software clients, etc.

Smart IP network protection
The IP PBX SMG-500 has intelligent protection against unauthorized external connections of SIP subscribers (dynamic firewall, static firewall, white/black lists, etc.) and connections via http/https/telnet/ssh.

High quality voice processing
The high quality of voice processing is provided by the up-to-date hardware platform, support for main audio codecs used in VoIP networks (G.711, G.726, G.729), echo cancellation, silence detector, comfort noise generator, receiving and generating DTMF signals, as well as traffic prioritization mechanisms (QoS).

Use Case

use case SMG-500.jpg

1 Optional
Documents and files
  • 4 × Е1 ports (RJ-48)
  • 4 × Ethernet 10/100/1000BASE-T ports (RJ-45)
  • 1 × USB 2.0; 1 x USB 3.0
  • 1 SD card slot (SDHC)
  • 1 COM port (RS-232, RJ-45)
VoIP protocols
  • H.323
Advanced SIP/SIP-T/SIP-I functions
  • SIP, SIP-T/SIP-I interaction
TDM protocols
  • SS7
  • PRI (Q.931)
  • Q.699 (PRI and SS7 interaction)
Voice codecs
  • G.711 (a-law, μ-law) 
  • G.726
  • G.729 (A/B) 
  • OPUS
  • AMR1
Voice standards
  • VAD (Voice Activity Detection)
  • CNG (Comfort Noise Generation)
  • AEC (echo cancellation, G.168 recommendation)
  • Interactive Voice Response (IVR) system with graphic editor
  • Direct Inward System Access (DISA) system
  • Call queue:
    • Various algorithms for choosing operators
    • Call distribution considering repeated client requests
  • Reporting system by operators/groups of operators (processed calls, missed calls, average waiting time, etc.)
  • Phone book:
    • Creating a phone book from the station subscribers list
    • Transferring a phone book to subscribers via LDAP
    • Obtaining a display name from the LDAP server
  • Video processing:
    • Transmitting a video stream using Video Offroad mode
Call management
  • Number modifications before and after routing
  • Call recording according by parameters
  • Routing by access category
  • Subscriber lines restriction
  • Subscriber service mode configuration
  • Trunk group cut-off
  • Direct connection of trunk groups
  • Prefix for multiple trunk groups
  • Limiting the number of simultaneous calls to the SIP interface
  • Ingress load limiting (call per second) for a trunk group
  • Interaction with the STUN server on a SIP server
  • Routing by Called Party Number (CdPN) and/or Calling Party Number (CgPN)
Quality of Service (QoS)
  • Diffserv assignment for SIP
  • Diffserv assignment for RTP
  • Transmission via INBAND, RFC 2833, SIP INFO, SIP NOTIFY
Supplementary Services
  • Call Forwarding:
    • Call Forwarding on Out of Service (CFOS)
    • Call Forwarding on No Reply (CFNR)
    • Call Forwarding Unconditional (CFU)
    • Call Forwarding on Busy (CFB)
    • Call Forwarding on Time (CFT)
  • Call Transfer
  • Music on Hold (MOH)
  • Call Hold
  • SIP-forking support for SIP subscribers
  • Call Hunt
  • Call Pickup
  • Call Parking
  • Busy Lamp Field
  • Add-on conference (CONF)
  • Conference based on subscribers list
  • 3-Way conference
  • Intercom
  • Paging Call
  • Call Queue
  • Сall Back when the position in queue is reached1
  • Call Recording
  • PINCodeAccess
  • Follow me
  • Follow me on No Response
  • Do not disturb (DND) with whitelist
  • Blacklist
  • Intervention
  • Voice mail
  • One Touch Record
  • Uploading/downloading configuration as a single file
  • Creating multiple network interfaces for telephony (SIP, RTP) with different IP addresses
  • Operation with multiple dial plans
  • SS7 signal channel backup
  • Voice activity control (by the presence of RTP or RTCP)1
Management and monitoring
  • E1 and VoIP channels monitoring in web interface
  • Channels and SS7 signal links management via web interface
  • Alarm logging with the option of storing entries on the syslog server
  • Storing traces on SD card/USB storage device
  • Alarm reporting via SNMP
  • Automatic logging activation after gateway restart
  • Monitoring of active sessions of the web interface users
  • Billing data is recorded to CDR file. Simultaneously, CDR file is recorded to a local SD disk, USB storage device or remote FTP server
  • RADIUS Accounting
  • Supported billing systems:
    • Hydra Billing
    • LANBilling
    • PortaBilling
    • NetUP
    • BGBilling
  • Integration with other systems is possible
  • Black and white IP addresses lists for registration
  • Logging of all access attempts to the device
  • Automatic IP blocking after unsuccessful login attempts or/and access via http/https/telnet/ssh
  • List of IP addresses allowed to manage the device
  • Multilevel web interface access permission
  • SIP subscribers authentification
  • RADIUS authorization (RFC 5090, Draft-Sterman)
Physical specifications and ambient parameters
  • Operating temperature: from 0 to +40 °С
  • Relative humidity: up to 80 %
  • Power supply: 220 V AC +-20%, 50 Hz
  • Lead-acid battery: 12 V
  • Battery charge current: 1.6+-0.1 A
  • Low battery voltage threshold indication: 11 V
  • Threshold voltage for battery deep discharge protection: 10-15.5 V
  • Power consumption: up to 40 W during battery charge, up to 20 W without battery charge
  • Dimensions (W × H × D): 430 × 44 × 203 mm
  • Implementation: 19", 1U
  • Weight: 2.35 kg

¹Not supported in the current firmware version

Operational lifetime of the ELTEX equipment
In development
Mass production
Mass production is over
Sold out
Support is over
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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