ANALOG VOIP MEDIA GATEWAY

SMG1000-D16S16O

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

Analog Overview

SMG1000-D96S/D120S/D144S

HomeSMG1000-D16S16O

Hybrid FXS/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/FXO 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 analog FXS/FXO 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-D16S16O 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-D16S16O 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/FXO, 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-D16S16O) 32 FXS/ FXO channels (SMG1000-D16S16O)
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	LWH 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

Session border controller ( SBC)

Analog VoIP media gateway

Digital VoIP media gateway

IP-PBX & multimedia platform

Call recording cards & solution

HMP/VOIP/TDM cards & platform

ANALOG VOIP MEDIA GATEWAY

SMG1000-D16S16O

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