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ANALOG VOIP MEDIA GATEWAY

SMG1000-D32S

SMG1000-D32S

32Ports FXS Analog VoIP Media Gateway, Cost Effectively And Reliably Connect Your Legacy Networks to VOIP

  • 32 Ports FXS to SIP
  • High Price-Performance Ratio
  • Custom Made Auto-Provision
  • DSP Media Processing
Maximize and Migrate Value of Existing PSTN Networks To VoIP

With a simple and economical way to help legacy telephone, fax machine and PBXs interconnect with IP network, Synway's 32 ports analog FXS gateway enables call center and multi-branch enterprises to process powerful, versatile and efficient VoIP solutions with unparalleled cost advantages. Connected between a PBX, LAN or WAN, the 32 ports FXS VoIP Gateway converts analog PSTN messages into a format suitable for transmission over standard IP networks.

Rich Customizable Features and Robust Telco' architecture

Designed for voicemail and unified messaging applications, the 32 Ports Analog VoIP Gateway SMG1000-D32S has a 10/100/1000M (optional) Base-T Ethernet connection for connecting legacy PBX to a LAN. The analog loop start functionality supports integration via in-band signaling (DTMF or FSK), serial protocols, as well as T.38 for fax transmissions over IP (FoIP).

Rich Customizable Features and Robust Telco' architecture

Designed for voicemail and unified messaging applications, the 32 Ports Analog VoIP Gateway SMG1000-D32S has a 10/100/1000M (optional) Base-T Ethernet connection for connecting legacy PBX to a LAN. The analog loop start functionality supports integration via in-band signaling (DTMF or FSK), serial protocols, as well as T.38 for fax transmissions over IP (FoIP).

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Features
Completely non-blocking architecture and Scalable System
Easy integration with existing telephony interfaces
Open-standard SIP support and register to multiple SIP proxy servers
Make and receive IP calls from analog extensions
Call budgeting based on allocated amount, minutes and call count
Manageable based call routing TDP-IP/IP-TDM
Restrict unwanted calls with list of denied numbers
Real-time call record send to CDR server
Caller ID presentation and restriction
Hotline extension setting
Web-based remote administration
Consol access via Telnet, SSH.
Capabilities
High performance VoIP connectivity for SMBs
Voice optimization to ensure better user experiences
Enhanced call routing ability with high voice quality
Easy to install, configure, and maintain
Support IPv4 and IPv6 international network
Data/voice/management VLAN and more
Build-in firewall and access rules
Support SNMP/TR069/Auto-Provision
Cloud-based management and bandwidth optimization
Support SIP, MGCP or other customizable protocols
Primary/Backup SIP Servers
Flexible routing and manipulation
Core Specifications
Physical Interface Phone Interface: 32 Ports FXS, RJ-11(some models are RJ21 available)
Ethernet Interface: 2* RJ-45 10/100Mbps Base-T Ethernet, Female RJ-45
1000M LAN/WAN available for some product models while required
Session Capacity 32 SIP channels (SMG1000-D32S)
32 FXS channels (SMG1000-D32S)
Connectivity SupportedDial Mode: DTMF and Pulse
Pulse: 10 and 20PPS
Caller ID:DTMF/FSK
Max Cable Length:5KM
Reversed Polarity
OpenVPN
VoIP Protocols TLS / SRTP ;OpenVpn;SIP V2.0 (RFC 3261,3262,3264);IMS/3GPP; SDP ;REFER (RFC 3515);RTP/RTCP;STUN  (RFC3489) ;ARP/RARP (RFC 826/903);SNTP (RFC 2030);DHCP/PPPoE;TFTP/HTTP/HTTPS;DNS/DNSSRV (RFC 1706/RFC2782);VLAN802. 1P/802.1Q
Call & Routing

T.38/Pass-through, up to 14.4kbpsPort Groups
IP Trunks
Primary and Secondary SIP Account
32 Inbound/Outbound Routing
Number Manipulation
Digit maps
TDM to IP or IP to TDM
IP load balancing
IP fault tolerance

Voice Capability

G.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMRG.711A/U law, G.723.1, G.729A/B,G.726,iLBC,AMR
Comfort Noise Generation(CNG)
Echo Cancellation(G.168)
DTMF mode: Signal/RFC2833/INBAND
Silence suppression with comfort noise
G.168 automatic echo cancellation
Call Progress Analysis (CPA), including Positive Voice Detection, Positive Answering
Machine Detection (PAMD), DTMF detection, and fax tone detection
Manageable based call routing TDP-IP/IP-TDM.
Restrict unwanted calls with list of denied numbers.
Voice Activity Detection (VAD)
Adaptive (Dynamic) Jitter Buffer
Programmable Gain Control
Hook Flash

FoIP Protocol & Faxing

Static IP, PPPoE, DHCP ClientT.38 for transmission over a packet network
T.38/Pass-through, up to 14.4kbps
T.38 FoIP: transcode fax from T.30 fax protocol (supporting V.17) modulation schemes

Network Capability Static IP, PPPoE, DHCP Client
IPv4, IPv6
Static/dynamic ARP
DIFFServ, ToS
NAT (Rout and Bridge)+
MAC Address Clone
Static routing+
Built-in Firewalls
QoS, Traffic Shaping
Voice/Data/Management Vlan
Maintenance & Upgrading SNMP/TR069.
Auto Provision
Action URL
Digit map
Web/Telnet. ACL
Configuration Backup/Restore
Bandwidth Optimization
Routing Rules based Prefixes
Firmware Upgrade via WEB
Syslog and CDR.
Access Rule list.
Network Capture
Outward Test(GR909).
Automatic Time Synchronization
IVR local Maintenance.
Cloud-based Management
Caller/Called Number Manipulation
Open-standard SIP support and register to multiple SIP proxy servers.
Application Capabilities SupportedDial Mode: DTMF and PulseCall waiting
Blind Transfer
Attend Transfer
Call forward on Busy
Call forward on No Reply
Unconditional Call Forward
HotlineCall hold
DND
Call Pickup
3-way conference
Voicemail
Conferencing Resource Call budgeting based on allocated amount, minutes and call count
Complete non-blocking architecture and Scalable System
Hotline extension setting
Support 3-Way and Multi-Way Conferencing
Environment & Power Power Supply: 100-240V, 50-60Hz+
Power Consumption: Approximately 50W
(Storage): -20 ~85°C
Humidity: 10%-90% No condensation.
Operating temperature range: -10 ℃ ~ 55 ℃
Physical Dimension

L*W*H 440(mm)*202(mm)*44(mm)
Weight Approximately 5.95ibs(about 2.7kg)

Warranty/Certifications 3 years: The first year exchange for free. On the Second & Third free to repair.
CE, FCC or Any other Certificates Customizable
Broadsoft, Elastix, Asterisk, Teams and other UC platform 
Resources
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Analog Overview
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Success Stories & Applications
Synway VoIP Gateways Enable Smooth Integration between Legacy TDM and IP Contact Center Platforms
ChinaUnionPay selected Synway' SMG series VoIP Gateway to connect its TDM and IP-based contact center operations to the PSTN.
ChinaUnionPay selected Synway's VoIP Gateway to connect its TDM and IP-based contact center operations
Asterisk based IP-PBXs are becoming more available and popular on the PBX business market, threatening the traditional proprietary PBX manufacturers position continuously.
Global BPO Provider Huayun Data Chose Synway SMG Gateway for Multiple Offshore Service Center
Founded in 1998 and headquartered in USA, Huayun Data, a leading provider of global Business Process Outsourcing (BPO) services, deployed Synway SMG2000 media gateways at its locations in the US to enable a one-stop, cost-effective and reliable VoIP connections between its affiliated agencies.......
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