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Matrix Telecom SETU VTEP - User Manual

Matrix Telecom SETU VTEP
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SIP Trunking with Elastix
Configuration Guide for Matrix SETU VTEP
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Overview

The Matrix SETU VTEP is a versatile telecommunication device designed to facilitate SIP trunking with Elastix, offering a robust solution for integrating traditional E1/T1 networks with modern IP-based communication systems. This device acts as a gateway, bridging the gap between legacy PBX systems and VoIP platforms, thereby enabling organizations to leverage the cost-effectiveness and flexibility of SIP-based telephony while retaining their existing infrastructure. Its primary function is to convert E1/T1 signals into SIP packets and vice versa, allowing for seamless call routing and communication between different network types.

Function Description

At its core, the Matrix SETU VTEP serves as an E1/T1 to SIP gateway. This means it can terminate E1/T1 lines from a traditional PBX or the PSTN and convert the voice traffic into SIP (Session Initiation Protocol) for routing over an IP network, typically to an Elastix server. Conversely, it can receive SIP calls from the Elastix server and convert them back into E1/T1 signals for delivery to the PBX or PSTN. This dual-direction conversion capability is crucial for organizations looking to migrate to VoIP incrementally or maintain a hybrid communication environment.

The device supports SIP trunking, which involves establishing a virtual connection over the internet to carry voice calls. By configuring a SIP trunk between the SETU VTEP and an Elastix server, calls originating from the E1/T1 network can be routed through Elastix, benefiting from its advanced features such as IVR, call queues, and voicemail. Similarly, calls initiated from extensions registered with Elastix can be routed through the SETU VTEP to the E1/T1 network, allowing them to reach external landlines or other PBX extensions.

The SETU VTEP is designed to handle various call scenarios, including outgoing and incoming calls between the E1/T1 and SIP networks. For outgoing calls from Elastix, the device is configured to recognize specific dial patterns and route them through the E1/T1 port. This allows Elastix users to make calls to the traditional telephone network. For incoming calls from the E1/T1 network, the SETU VTEP directs them to the Elastix server, which then routes them to the appropriate extension or service based on its internal configuration.

Beyond basic voice calls, the SETU VTEP also supports FAX communication. It can be configured to handle FAX protocols, specifically T.38 (UDPTL) or Pass-through, ensuring that fax transmissions are reliably carried over the IP network. This is a critical feature for businesses that still rely on fax for certain operations, as it allows them to integrate fax services into their VoIP infrastructure without needing separate analog lines.

The device's architecture is built to ensure robust and reliable communication. It manages the signaling and media conversion processes, ensuring that voice quality is maintained and calls are established and terminated efficiently. The integration with Elastix is a key aspect, as Elastix provides the intelligence for call routing, user management, and advanced telephony features, while the SETU VTEP handles the physical layer connectivity and protocol conversion.

Usage Features

The Matrix SETU VTEP offers a user-friendly web-based Graphical User Interface (GUI) for configuration and management, making it accessible to administrators without requiring specialized command-line expertise. This GUI allows for comprehensive setup of all device parameters, from basic network settings to intricate SIP trunk and E1/T1 routing configurations.

One of the primary usage features is the ease of SIP trunk configuration. Administrators can define SIP trunks by assigning a name, specifying the SIP ID, and configuring registrar settings. The device supports peer-to-peer SIP trunking, which simplifies the setup by directly connecting to the Elastix server's IP address. This direct connection ensures efficient call routing and reduces potential points of failure. The GUI provides clear fields for entering the Elastix server's IP address as the destination for SIP calls, ensuring that the SETU VTEP knows where to send incoming SIP traffic from the E1/T1 network.

The SETU VTEP also offers flexible codec preference settings. Users can select and prioritize various voice codecs, such as G.711 A-Law and G.711 μ-Law, to optimize voice quality and bandwidth usage. The ability to move codecs between "Available" and "Selected" lists, and to set their preference order, allows administrators to fine-tune the audio performance according to their network conditions and requirements. This is particularly important for ensuring clear communication over IP networks, where bandwidth can be a limiting factor.

Handling of incoming and outgoing calls is another key usage feature. For incoming calls from the E1/T1 network, the SETU VTEP can be configured to route them to a specific destination on the SIP network, typically an Elastix extension or an IVR. This allows for precise control over how calls entering the system are handled. For outgoing calls from the SIP network, the device can be configured to recognize specific dial patterns (e.g., numbers prefixed with '1') and route them through the E1/T1 port to the traditional telephone network. This enables Elastix users to make external calls seamlessly.

The device also includes features for managing call progress and signaling. For instance, it can be configured to connect the source port when a 183 (Session Progress) message is received on SIP, which helps in handling early media and ensuring proper call setup. Similarly, for E1/T1 calls, it can connect the source port when a progress indicator is received, ensuring that call progress tones are correctly passed through.

FAX configuration is straightforward, allowing administrators to select between T.38 (UDPTL) and Pass-through protocols. This flexibility ensures compatibility with various fax machines and services, enabling reliable fax transmission over the IP network. The GUI provides clear options for selecting the desired fax protocol, simplifying the setup process.

The web GUI also provides a clear overview of the device's status, including the status of configured SIP trunks. This allows administrators to quickly check if a SIP trunk is active or disabled, and to view registration times and retry counts, which are useful for troubleshooting and monitoring the health of the communication system.

Maintenance Features

The Matrix SETU VTEP is designed with several features that aid in its maintenance and monitoring, ensuring continuous and reliable operation.

One of the most important maintenance features is the Status page available through the web GUI. This page provides real-time information about the device's operational state. Specifically, the "SIP Trunk Status" section allows administrators to view the status of each configured SIP trunk. This includes whether a trunk is "Active" or "Disabled," the "Registration Time," and the "Registration Retry Count." This information is invaluable for diagnosing connectivity issues with the Elastix server or other SIP endpoints. If a trunk is not active, the retry count can indicate persistent problems, prompting further investigation.

The web GUI itself serves as a primary maintenance tool, allowing administrators to access and modify configurations remotely. This eliminates the need for physical access to the device for most maintenance tasks, making it convenient for systems deployed in various locations. The ability to submit changes and have them applied immediately, often with a system reload, ensures that configuration updates can be performed efficiently.

The device's configuration guide, which outlines the steps for setting up SIP trunks, call routing, and fax services, also acts as a maintenance resource. It provides clear instructions for initial setup and for making adjustments as network requirements change. This documentation helps administrators understand the device's capabilities and how to configure them correctly, reducing the likelihood of misconfigurations that could lead to service disruptions.

In terms of software, the device's reliance on a stable CentOS base (as indicated by the server OS information) suggests a robust and well-supported operating environment. While the manual focuses on configuration, the underlying software platform typically benefits from regular updates and security patches, which are crucial for long-term system health. Although the manual does not detail the update process for the SETU VTEP itself, the mention of software versions implies that updates are part of its lifecycle.

The ability to configure various parameters, such as maximum channels and different call hold methods (RFC3261, RFC2543), allows for system optimization and troubleshooting. If call capacity issues arise, administrators can review and adjust these settings. Similarly, if there are problems with call holding, the ability to select different methods provides flexibility in resolving such issues.

The "Apply Configuration Changes Here" mechanism in Elastix, which reloads the configuration after changes are submitted, is also a maintenance-related feature. It ensures that any modifications made to the SIP trunk or routing rules are immediately put into effect, allowing administrators to test changes and verify their impact without significant delays.

Furthermore, the device's support for different DTMF modes (rfc2833, Outband) is a maintenance consideration. If there are issues with touch-tone recognition, administrators can switch between these modes to find the most compatible setting for their environment, ensuring that IVRs and other touch-tone services function correctly.

Overall, the Matrix SETU VTEP is designed to be a manageable and maintainable device, with its web-based GUI, clear status indicators, and flexible configuration options contributing to its operational longevity and ease of support.

Matrix Telecom SETU VTEP Specifications

General IconGeneral
ProtocolsSIP, H.323
TypeVoIP Gateway
Weight3.5 kg
Operating Temperature0°C to 45°C
Power SupplyAC 100-240V, 50/60Hz
Humidity10% to 90% non-condensing