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Synway FXO / FXS VOIP Gateways

Synway FXO, FXS Analogue to VOIP gateways allows you to connect your traditional analog phones, fax machines and PSTN lines in to VOIP systems such as Asterisk, Sangoma and 3CX. FXO Gateways are using to connect Analog PSTN lines coming from your Telco provider in to your VOIP system. FXS gateways are using to connect traditional analog phones and Fax Machines in to VOIP System or VOIP enabled IP PBX System. Your can register SIP extensions created in your IP PBX system on FXS gateway and assign to each analog port.

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With a simple and economical way to helps legacy telephone, fax machine and PBXs interconnect with IP network,Synway SMG1000 analog media gateway enables call center and multi-branch enterprises to process powerful, versatile and efficient VoIP solutions for unparalleled cost advantage.

Synway SMG 1000 series voice gateways are excellent quality and high cost-effective products that can connect legacy telephones, FAX, POS and PSTN lines. With carrier-grade performance,  SMG 1000 Series voice gateways provide flexible networking for large enterprises and SMEs. Easier installation & confuguration, TP series are compatible with IP PBX or soft switch such as Asterisk, 3CX, Avaya, Cisco, Huawei, Broadsoft, ZTE, etc.

Key Features:

Voice over Internet Protocol (VoIP) – Supports SIP per RFC 3261. Uses Real-time Transport Protocol/Real- Time Control Protocol (RTP/RTCP) for delivery of voice over the LAN
or WAN

Enhanced voice processing – Supports a variety of compression algorithms, including G.711 A-law and ?-law, G.729AB

T.38 Fax over Internet Protocol (FoIP) – Transcode fax from T.30 fax protocol(supporting V.17) to T.38 for transmission over a packet network

Web server interface – Each gateway unit is delivered with a web server interface, allowing configuration and software upgrades via a web browser



Physical InterfacePhone Interface: 4-32 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 Capacity4-32 SIP channels
4-32 16 FXS / FXO channels
ConnectivitySupportedDial Mode: DTMF and Pulse
Pulse: 10 and 20PPS
Max Cable Length:5KM
Reversed Polarity
VoIP ProtocolsTLS / SRTP ;OpenVpn;SIP V2.0 (RFC 3261,3262,3264);IMS/3GPP; SDP
;REFER (RFC 3515);RTP/RTCP;STUN (RFC3489) ;ARP/RARP (RFC 826/903);
(RFC 1706/RFC2782);VLAN802. 1P/802.1Q
Call & RoutingT.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 CapabilityG.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 & FaxingStatic 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 CapabilityStatic IP, PPPoE, DHCP Client
IPv4, IPv6
Static/dynamic ARP
NAT (Rout and Bridge)+
MAC Address Clone
Static routing+
Built-in Firewalls
QoS, Traffic Shaping
Voice/Data/Management Vlan
Maintenance & UpgradingSNMP/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 CapabilitiesSupportedDial Mode: DTMF and PulseCall waiting
Blind Transfer
Attend Transfer
Call forward on Busy
Call forward on No Reply
Unconditional Call Forward
HotlineCall hold
Call Pickup
3-way conference
Conferencing ResourceCall 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 & PowerPower Supply: 100-240V, 50-60Hz+
Power Consumption: Approximately 50W
(Storage): -20 ~85°C
Humidity: 10%-90% No condensation.
Operating temperature range: -10 ? ~ 55 ?


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