Key Features
Featured List

Modern organizations are recognizing VoIP as the cost-effective alternative to the wide spread ISDN networks. Enterprises are inclined towards availing benefits of IP networks while protecting their existing investment against changing communication technologies. They seek solutions to avail benefits of both IP and ISDN trunking through a single device to optimize operational expenses and save more on long distance calls.

Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.

Key Features

Allowed and Denied Number Lists

Allowed and Denied Lists are used to restrict dialling of long-distance numbers. A number is blocked if its prefix matches any entry in the Denied Lists. On the other hand, a number is allowed to go through if matched with any entry of Allowed List. This provides flexibility of allowing only specific numbers while blocking all others.

Automatic Number Translation (ANT)

SETU VTEP supports simultaneous registration of multiple SIP accounts. These accounts can be availed from single or multiple ITSPs. While placing a call, a caller is not conscious of the routing logics defined and the SIP account in use. The gateway itself modifies the dialed number or part thereof so that it matches with the numbering plan that is understood by the ITSP whose trunk is used for the specific call.

Call Detail Records (CDR)

SETU VTEP can store details of 2000 calls in its memory. Call reports can be generated using filters like source port, destination port, calling number, called number, date, time and duration.

Caller-ID Based Routing

Based on the Caller-ID details, an incoming call can be routed to a pre-defined port. For example, important business calls may be directed to the higher authorities, calls with specific Caller-ID may be directed to specific departments, while calls from anonymous numbers may be directed to the customer support teams. Users with Digital key phone or IP phone can also have display of the caller’s name, if programmed accordingly.

Call Progress Tones and Rings

The gateway offers flexibility of programming call progress tones and ring cadence to match the standards of the country of installation. Country Specific Call Progress tones like Dial Tone, Ring Back Tone, and Busy Tone etc. can also be programmed.

Digest Authentication

Digest authentication allows an incoming call on the gateway to be screened using encrypted authentication keys. Only on successful authentication, the gateway allows the call to be established. The automated security mechanism allows restricting malicious calls. Calls received are thereby restricted to a number of trusted callers.

Dynamic DNS (DDNS)

The compact gateway comes with an embedded Dynamic DNS client that automates the discovery and registration of its IP addresses on the public network. The remote administrator and peer-mode devices can thereby connect to the gateway using the Domain Name associated with the dynamic IP. The gateway can thereby seamlessly establish calls even when allocated a dynamic IP.

Fax over IP (FoIP)

SETU VTEP can be used to send and receive Fax using a SIP account, over the IP network. The gateway supports FoIP using T.38 Vocoder and Pass-Through. Sending Fax over Internet eliminates the need of dedicated analog lines to send fax over long distances.

Multiple SIP Accounts

32 SIP accounts can be programmed and each call placed can use a specified trunk. Dynamic allocation of SIP account is also possible via dial plans, based on various call routing algorithms.

Network Clock Synchronization

Whenever the PBX system is interfaced between ISDN T1/E1 line and a communication device like VoIP gateway, chances of clock slip exists as PBX and gateway operate on different clock sources. As a consequence of clock slip, many operational and integration problems like poor voice quality and noisy or dropped calls occur during field installations.
Traditional ISDN network provide extremely accurate clock signals. SETU VTEP is equipped with SYNC IN and SYNC OUT ports that acquires stable network clock and uses it to provide clock synchronization with the attached PBX system, eliminating any synchronization issue and noisy or dropped calls.

Peer-to-Peer Calling

SETU VTEP supports VoIP calls between two locations without going through a proxy server. IP addresses of the locations can be programmed in its Peer-to-Peer table. 500 such entries can be programmed. Short, numeric dialing codes can be defined for calls between these locations. With the embedded Dynamic DNS client, calls can also be established between devices on dynamic Public IP. Organizations having multiple locations like branch offices and factories can use this feature to provide direct dialing between these end-points. Since the Peer-to-Peer calls are placed over the public IP network, the call cost is minimal.

Return Call to Original Caller (RCOC)

A call attempt may be unsuccessful if the called party is busy, does not pick up the call or due to any network issues. SETU VTEP logs such unsuccessful calls in a RCOC table with details about the caller, number dialled and time of call. With these details available, if a call back is received from any of the called number, it is possible to route the call to the specific caller who attempted a call to the concerned number. This greatly reduces communication delay and also eliminates the need for operator assistance to redirect the call.

Remote Programming

Matrix SETU VTEP incorporates built-in HTTP server and web pages for easy configuration. This Web based programming feature allows a user to configure the gateway from any part of the world, once connected to the IP network.

Quality of Service (QoS)

Layer3 QoS prioritizes voice packets over IP network, as voice traffic is delay sensitive. Toll-quality echo cancellation for a tail length of 128ms can be done on a per channel basis. CNG, in conjunction with VAD algorithms, quickly detects absence of audio and conserves bandwidth by preventing the transmission of these silent packets.

VLAN Tagging

A Virtual Local Area Network (VLAN) may be defined as a group of LANs that have different physical connections, but which communicate as if they are connected on a single network segment. With VLAN tagging, the gateway receives packets only tagged for it. The packets sent by the gateway are also given specific tag. VLANs increase overall network performance by grouping users and resources that communicate most frequently with each other.
Featured List
Allowed and Denied Numbers MAC Cloning
Automatic Number Translation NAT and STUN Support
Call Detail Record PCAP Trace
Caller Line Identification and Presentation (CLIP) Peer-to-Peer Calling Table
Caller-ID based Routing PIN Authentication
Day Light Saving Mode Programmable Access Codes
Date and Time Settings Programmable Call Progress Tones
Digest Authentication Remote Held and Transfer
Direct Dialing-In (DDI on T1/E1/PRI) Return Call to Original Caller
Dynamic DNS System Log Client
Emergency Number Dialing VLAN Tagging
Fax over IP (T.38 and Pass-Through) Web based Programming