Enterprise IP PBX SMG-200
Mass production
Key Benefits
- IP PBX for up to 200 subscribers
- Up to 50 calls simultaneously
- Up to 16 FXS/FXO ports
- 4 LAN ports
- Call center functions
- Call recording
SMG-200 is an enterprise PBX for 200 subscribers with a full set of Supplementary Services (SS).
The basic configuration of the enterprise IP PBX SMG-200 is designed to connect up to 100 SIP subscribers and can be extended to connect up to 200 subscribers when purchasing the appropriate software1. 16 RJ-11 ports can be used to connect analog telephones as well as for connection to land lines. LAN ports are designed for connection to carriers' networks using SIP trunks, as well as to connect VoIP gateways (e.g., TAU-24 supporting for 24 FXS ports) to increase the number of FXS/FXO ports. Call recordings and CDR files are stored on the SD card or USB drive. It is also possible to automatically upload files to an FTP server.
Networking of separated offices
The SMG-200 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-200 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-200 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-200 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).
1Optional
The basic configuration of the enterprise IP PBX SMG-200 is designed to connect up to 100 SIP subscribers and can be extended to connect up to 200 subscribers when purchasing the appropriate software1. 16 RJ-11 ports can be used to connect analog telephones as well as for connection to land lines. LAN ports are designed for connection to carriers' networks using SIP trunks, as well as to connect VoIP gateways (e.g., TAU-24 supporting for 24 FXS ports) to increase the number of FXS/FXO ports. Call recordings and CDR files are stored on the SD card or USB drive. It is also possible to automatically upload files to an FTP server.
Networking of separated offices
The SMG-200 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-200 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-200 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-200 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
1Optional
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Specifications
Documents and files
Warranty
Interfaces
1Not supported in the current software version
- 16 × FXS/FXO RJ-11 ports
- 4 × Ethernet 10/100/1000BASE-T ports (RJ-45)
- 1 × USB 2.0; 1 × USB 3.0
- 1 SD card slot (SDHC)
- 1 COM port (RS-232, RJ-45)
- SIP, SIP-T/SIP-I
- H.323
- SIP, SIP-T/SIP-I interaction
- G.711 (a-law, μ-law)
- G.726
- G.729 (A/B)
- OPUS1
- AMR1
- VAD (Voice Activity Detector)
- CNG (Comfort Noise Generation)
- AEC (Acoustic Echo Cancellation, G.168 recommendation)
- AGC (Automatic Gain Control) (for FXS/FXO ports)
- 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.)
- Pulse and tone dialing
-
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
- Number modification before and after routing
- Call recording 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 via SIP interface
- Dialing rules for FXO ports
- Hotline for FXS ports
- Routing by Called Party Number (CdPN) and/or Calling Party Number (CgPN)
- Diffserv assignment for SIP
- Diffserv assignment for RTP1
- 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
- AutoCLIP for FXO ports
- 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
- PIN Code Access
- Follow me
- Follow me on No Response
- Do Not Disturb (DND) with whitelist
- Blacklist
- Calling Line Identification Presentation (CLIP) in FSK formats (ITU-T V.23, Bell 202), DTMF, “Russian CLI”
- Caller ID and starting time of a call issuance in FSK mode
- Calling Line Identification Restriction (CLIR) (for FXS ports)
- Intervention
- Voice mail
- One Touch Record
- Speed Dial for FXS ports
- Uploading/downloading configuration as a single file
- Creating multiple network interfaces for telephony (SIP, RTP) with different IP addresses
- Operation with multiple dial plans
- Voice activity control (by the presence of RTP or RTCP)1
- Alarm logging with the option of storing entries on the syslog server
- Storing traces on SD card/USB storage device
- Alarm reporting via SNMP
- FXS lines testing and monitoring
- Telephone connection display (FXS)
- 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 address 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)
- 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: 2.47 kg
1Not supported in the current software version
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Descriptions
Software
Documentations
Certificates
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
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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