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E1 to SIP Converter

Dinstar -MTG600

MTG600 1 port E1 US$ 1,494

MTG600 2 port E1 US$ 1,793

MTG1000 1 port E1 US$ 1,776

MTG1000 2 port E1 US$ 2,153

MTG1000 4 port E1 US$ 4,111

MTG1000 8 port E1 US$ 7,540

Dinstar E1 Trunk Gateway cocok untuk Asterisk juga untuk PABX lainnya yang lebih luas.

MTG600 series is a kind of digital trunking gateway based on embedded operating system. It is not only support ASTERISK but also GENERAL E1 system. It supports standard SIP/H.323 protocol, with large-capacity carrier class telephone trunking gateway functions. Currently it supports 1/2 E1/T1 interfaces and can realize intercommunication with mainstream manufacturers softswitch system, and interwork with carrier's local Telephone exchange by PRI/SS7 interface.

The main function of MTG series digital trunk gateway:

 MTG is located in the edge of PSTN and IP and is enabling the delivery of SIP / H.323 into PSTN networks, as well as IP to IP transcoding for IP network peering applications. MTG configured with 1 to 32 interfaces per chassis. It completes intercommunication with softswitch through SIP/H.323 trunk; realize interconnection and intercommunication with traditional PSTN through SS7 or PRI;

 Support multiple voice codec of ITU-T stardand: G.711A，G.711μ，G.729A/B，G.723.1, etc;

 Advanced IP voice signal processing ensured the high quality assurance (QoS): Realtime Transport Protocol (RTP) voice packetizing and depacketizing technology, dynamic jitter buffer management, echo cancellation (ITU-T G.168/165 ), dynamic compensation for packet loss, Voice Activity Detection (VAD) technology and Comfort Noise Generation (CNG) technology;

 Support multiple management and maintenance methods, such as SNMP, COM port, and Telnet command line, etc;

The technical specifications

Technical parameters and specifications of MTG1000
Call control	H.323, SIP protocol, PRI/SS7 signaling
Voice processing	Voice code : G.711 64kbps，G723.1 5.3/6.4Kbps，G.729A/B 8Kbps Echo cancellation: G.168, up to 128ms Support Adaptive Voice Activity Detection (VAD) with Comfort Noise Generation (CNG), Packet Loss Concealment Support T.38 Support Pass-through
Data features	Support DNS client
Network management	Support SNMP Support TELNET for remote configuration and software upgrade
Billing	Radius+ Interface
Trunk interface	Provide 1~32 E1/T1 interfaces with RJ48/BNC
Ethernet	Auto-negotiating 10/100Base-TX Ethernet port.
Console interface	RS-232 COM port, 9600bps
Operating temperature	0～40℃
Storage temperature	-20～80℃
Operating Humidity	10%～90%
Dimensions	43.5cm (width, 19”) * 50.0cm (depth) * 4.45cm (height, 1U)
Power supply	85～265V AC，50～60Hz，1.2A
Power consumption	40W
Weight	6.5Kg

Wellgate-5260

1 ports E1 ( 30 channels) US$ 4,037

4 ports E1 ( 120 channels) US$ 10,110

Hardware dilengkapi dengan card merek AudioCodes yang terkenal dengan kualitas terbaik di kelas atas.

WellGate 5260 is an UNIVERSAL VOIP GATEWAY which navigates the calls in between H.323, SIP and PSTN freely , not simple PSTN to VOIP calls or vice verse. It can easily implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP and SIP, H.323 and H.323 calls simultaneously .With built-in PSTN/VOIP IVR helps service provider to establish their own voice service platform quick. It is suitable for application of Pre-paid Calling card service termination. The built-in both ways IVR provide flexible function for termination and Pin code authentication announcement.

WellGate 5260 provides an easy to use web-based managing interface. Administrator can use the web based drag and drop call-flow editor to design their VOIP service very quick and without losing any flexibilities. A web-based voice prompt management GUI is also provided to simplify the deployment of IVR related service.

WellGate 5260 provides three models as follows.

WellGate 5260-1, one line T1/E1 trunk gateway
WellGate 5260-2, two lines T1/E1 trunk gateway
WellGate 5260-4, four lines T1/E1 trunk gateway

It supports ISDN PRI, CAS, R2 and Q-SIG protocol over T1/E1 line to connect to PSTN switch or PABX.

Key Features

Navigate Call Freely in SIP, H.323 and PSTN
Support SIP RFC 3261 and ITU-T H.323 V5 Simultaneously
Up to 4 Programmable E1/T1 Trunks
PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG , SS7
Support Audio Codec G.711, G.723.1, G.729A, GSM
SIP to PSTN Call and vice versa
H.323 to PSTN Call and vice versa
H.323 to SIP Call and vice versa
SIP to SIP Call
H.323 to H.323 Call
Support up-to 16 Multiple SIP Proxy Servers
Support SIP Proxy, Gatekeeper and P2P Calls Simultaneously
Built-in Universal VOIP Address Book
Support Early Media and SIP Delay Media
Support RADIUS Authentication, Authorization and Accounting
Intelligent PSTN Call Routing and in-Trunk Hunting
Support Flexible VOIP Routing and Account Code
Flexible Digit Manipulation Plan
Support Calling/Called Number Replacement
In-band and out of Band DTMF Transmission
T.38 Fax Relay up to 14400 bps
Dynamic Call Treatment Based on Drag and Drop Call Flow Editor
Built-in PSTN and VOIP IVR
Provides Call Detail Record
Full Web Management Interface
Comply with RoHS
Support up-to 16 SIP Register/Outbound Proxy Servers
Support IP Access Control List with Account Code
Support IP, ANI, DNIS ACL Group
Added VOIP Hunting Features
Support VOIP Hunting based on Priority & Weekday/Time
Support PSTN Line Hunting based on Time
Made in Taiwan