VoIP Distribution
OpenVox DGW1008

OpenVox DGW1008(R) with dual power inputs, enables the continued operation

OpenVox Communication Co., Ltd, a global leading provider of the best cost effective open source Asterisk® telephony hardware and VoIP Gateways, today announces to release a new digital VoIP gateway product line to the enterprise communication market with the best value guaranteed. To Ensure the system stability, the hardware comes with a redundant power supply as default configuration. Thus it will be a plus when implementing the digital gateways in the enterprise communication systems.

There are four E1/T1 Gateway models, the DGW-1001(R), DGW-1002(R), DGW-1004(R) and DGW-1008(R). There is one port on DGW-1001 supporting 30 channels most, two ports on DGW-1002 supporting 60 channels most, four ports on DGW-1004 supporting 120 channels and eight ports on DGW-1008 supporting 240 channels most. The “R” means that the device supports redundant power supply. It is developed with a wide selection of codecs and signaling protocol, including G.711A, G.711U, G.729A, G.722, G.723 and GSM.  It supports PRI protocol. With user-friendly GUI, users may easily setup their customized VoIP gateways and connect them to the existing telephone systems or simply build up a new one. Also secondary development can be completed through AMI (Asterisk Management Interface) with our digital gateways. The DGW-100XR series gateway will be 100% compatible with all kind of SIP servers, such as Asterisk, Askozia, Elastix, trixbox, 3CX, FreeSWITCH and VOS VoIP operating plattform. OpenVox T1/E1 Gateway has good processing ability and stability and we provides 1/2/4/8 T1/E1 interface for your choice.

“We’ve gained a great success in releasing our VoIP GSM gateways to the market right from the beginning of 2013. And the VoIP gateway market is much bigger than we imagine.” said Lin Miao, the CEO of OpenVox, “We received plenty of feedbacks and inquiries about new VoIP gateways supporting E1/T1 interface that’s already in our roadmap. I am excited to announce the release of our digital VoIP gateways. It will continue to satisfy our customers needs for their business communications.”

OpenVox DGW-100XR series T1/E1 Gateway is an open source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. This kind of gateway connects traditional telephone system to IP networks and integrates VoIP PBX with PSTN seamlessly. With friendly GUI, users may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).

The new digital VoIP E1/T1 gateways will be ready for worldwide shipping now. Please contact VoIPDistri.com your Voice over IP Distributor by new OpenVox DGW-100XR PRI Digital VoIP Gateway with redundant power supplies or you can simply email at info@voipdistri.com for any enquiries.

OpenVox offer 4 models with single and redundant power:

VoIPDistri.com – Leading Distributor of VoIP Telephony! Due to they long established experiences in the field of Analog-, GSM-, ISDN-networks and Voice over IP VoIPDistri.com able to offer you a comprehensive service for all Voice over IP requirements and network Solutions. VoIPDistri.com – Main OpenVox Distributor. VoIPDistri.com is OpenVox strategic partner since from 2009, see more at: http://shop.voipdistri.com or http://voip.world

Become Reseller

This way you can register quickly and easily via the following link to Voice over IP Reseller Partner-Access* and benefit from wide experience in Voice over IP! (* trade licence or certificate of registration necessary)

You will get special reseller conditions from VoIPDistri.com You will get special reseller conditions for complete OpenVox Gateway and Interface cards of VoIPDistri.com product Portfolio – click here!

reventix GmbH belongs to the leading enterprises in Germany that provides individual communication solutions by using Voice-over-IP (VoIP)reventix GmbH belongs to the leading enterprises in Germany that provides individual communication solutions by using Voice-over-IP (VoIP). reventix offer there customers full-service from one source. From conception to development and implementing, reventix provide everything modern mid-size businesses need in order to be one step ahead when it comes to unified communication.

The VoIP PBX EASY, uses a cloud and is therefore applicable for companies that meet all the requirements for an upmarket and feature-rich telephony.

Among the classical features, this telephone system is also equipped with complex time tables, the integration of mobile end units, video-telephony, text messages and other high-end-features.

reventix provide a brand-new 100% TLS/SSL bug-proof communication. By encrypting the signals (SIP) and the data (RTP) voice calls as well as video and text messages are secure against industrial spying. Moreover, all of reventix systems are stored in German high-security data centres.

Security, flexibility, future oriented technology and excellent services – these are the reasons why so many mid-size businesses rely on our solutions. By using encryption we provide these companies with a secured cloud communication and the capability to protect themselves from spying.

VoIP for Resellers
Accounts, Trunks, Cloud PBX

  • Enhance your portfolio and strengthen your customer retention through reventix partnership models.
  • By being a Whitelabel partner, you can offer all products generated by our VoIP-cloud to your customers using your name.

Unified Communications
API (Application Programming Interface)

The Berlin Reventix GmbH now offers its own API (Application Programming Interface), owned telecommunications solutions can develop on the platform of SIPbase.de with the developers and innovators.

Reventix thus has a UC API developed by which the functionality of a whole VoIP cloud can be accessed in real time for the first time. CEO Michael Kundt: “On the real-time interface, for example, complete call flows freely controlled and the current live status and information about extensions are displayed without running their own communication platform for this” at CeBIT in Hall 11, Stand B56 provides the take the first applications of the API live before.

The new beroNet Cloud Managed VoIP Gateway and can be equipped with up to 3 modules. The new Gateway has a SD Card Slot, new Firmware, 12 Ports (no T-Adapter nexessary), SBC Support, App-Store (DHCP-Server, VPN, PPPoE, Asterisk) and supports T1

The new beroNet Cloud Managed VoIP Gateway and can be equipped with up to 3 modules. The new Gateway has a SD Card Slot, new Firmware, 12 Ports (no T-Adapter nexessary), SBC Support, App-Store (DHCP-Server, VPN, PPPoE, Asterisk) and supports T1

beroNet, German manufacturer of Cloud Managed VoIP Gateways and Appliances presents at the CeBIT its new Gateway as well as its new Telephony Appliance.

The big talking point that the telecommunications industry is moving at the moment is especially the planned shut-down of ISDN to ALL IP. Sooner or later this could mean in the worst case, that calls could not be possible anymore via ISDN for owners of classic PBXs. One possible solution could be Medium or long-term the migration to VoIP. In this scenario a VoIP Gateway can simulate ISDN and the classic PBX can be furthermore operated with ALL IP. This can reduce costs and thus enhance the competitiveness.

beroNet provides with the via Cloud configurable VoIP Gateways and Appliances the necessary technology in order to fulfill the complex requirements for a successful restructuring. The beroNet Gateways can be used as an access technology for modern VoIP systems, as break-out solution for traditional telephone systems to VoIP providers or as an adapter between the different communication technologies.

At the CeBIT 2015 beroNet presents to 2 new products: the new beroNet Gateway and the new beroNet Telephony Appliance.

The new beroNet VoIP Gateway is Cloud Managed and can be equipped with up to 3 modules. It has an SD card slot and T1 support. More Features: 12 Ports (no T-Adapter necessary), App-Store (DHCP-Server, VPN, PPPoE, Asterisk) as well as SBC support. With its T1 support it is the ideal solution for the US market and for VoIP providers.

In addition beroNet introduces its new Telephony Appliance. It is equipped with the beroNet hypervisor which enables the installation of PBXs as virtual machines. Also other telecommunications applications such as Fax servers can be executed simultaneously on the same device. The new Appliance offers up to 8GB of RAM, hardware virtualization and 3 module slots. In addition, apps can be installed via Cloud (3CX, Asterisk, Faxserver) as well as Backups of already installed Images can be created and restored. The new Appliance is the perfect solution for IT and Telecommunication integrators and offers the ideal platform for next generation Software PBXs.

„As technological innovation leader in the sector of VoIP, Cloud Managed VoIP Gateways and Appliances beroNet is able to present on the world’s biggest IT fair even 2 new product innovations. With the beroNet App-Store we offer a variety of Software Upgrades for our approved Gateways“ –  says beroNet CEO Christian Richter.

Interested customers and reseller can buying their beroNet Gateway from German wholesalers : VoIPDistri.com – Voice over IP Distribution

VoIP is about convergence, saving telecom costs. However, these types of systems also create more inroads for attack & lose lot of money because of VOIP frauds & attacks. In this article, I will discuss a number of different ways your communications system can be breached and how it can be protected.

Is your IP Phone system a target for VOIP attacks?
Every year the number of PBX fraud victims increases dramatically. More and more companies are targeted by individuals who are looking to bring down or exploit the communications system. Some do it for fun and others for illicit profit, but the end result is always the same… It results in the telephone bill of average 5,000$ USD to 80,000$ per attack to your carrier!

The most vulnerable targets remain small-medium size businesses that are new to managing their own VOIP. They either don’t have the IT experience and staff to properly secure and maintain the network, or they’re unaware of the risks altogether having recently switched from a landline system. Whatever the reason, many networks are consistently left unprotected. By the time most companies realize that something is wrong with their phone expenses, it’s too late—the network security has been compromised.

Here is the article link which explain about the VOIP attacks: http://www.nytimes.com/2014/10/20/technology/dial-and-redial-phone-hackers-stealing-billions-.html?_r=0

Toll fraud losses are growing at rate faster than global telecom revenues.

Things to be considered

  • The law is clear, you are the only responsible for the security of your phone system and any charges generated from it.
  • You will pay on average 5,000$ USD to 80,000$ per attack to your carrier.
  • Downtime of your whole system is very common.
  • In some cases you will have to find a different carrier.

Let’s first discuss what steps you can take to protect your account from hackers.

What is STM and how it can help you secure your VOIP infrastructure?

You may be familiar with UTM – Unified Threat Management device, but have you come across an STM – SIP Threat Management device, that is used to protect the IP PBX and IP Phones/Telephony infrastructure from threats/attacks?

The STM – SIP Threat Management device, is installed in front of any SIP based PBX system or gateway and offers extra layers of Toll fraud losses are growing at rate faster than global telecom revenues.

Things to be considered

  • The law is clear, you are the only responsible for the security of your phone system and any charges generated from it.
  • You will pay on average 5,000$ USD to 80,000$ per attack to your carrier.
  • Downtime of your whole system is very common.
  • In some cases you will have to find a different carrier.

Let’s first discuss what steps you can take to protect your account from hackers.

What is STM and how it can help you secure your VOIP infrastructure?
You may be familiar with UTM – Unified Threat Management device, but have you come across an STM – SIP Threat Management device, that is used to protect the IP PBX and IP Phones/Telephony infrastructure from threats/attacks?

The STM – SIP Threat Management device, is installed in front of any SIP based PBX system or gateway and offers extra layers of security against numerous types of attacks that are targeted towards IP telephony infrastructure. The features offered by the STM complement those of a traditional firewall or UTM, and it can be installed in conjunction with a UTM.

Typical STM Installation Diagram
Here is a diagram of a typical STM installation in a VOIP network:

IP PBX Firewall (STM) - As a customer, you are responsible for securing your phone system. On average, an attack costs several thousands of US dollars. Our STM is installed in front of any SIP based PBX or gateway offering several layers of security against numerous types of attacks. Block specific IPs or countries, protect your PBX against hackers trying user names and passwords, someone is trying to flood your PBX with a DDos attacks? No problem!

Overview of the most common attacks to PBXs today and how the ALLO STM handles them

  1. SIP Device Fingerprinting: The hacker will try to identify which PBX software is running or which hardware you are using. Once he gets this info, he will look for their weaknesses and attack accordingly. The STM will simply not answer to such requests leaving the hacker in the dark.
  2. User enumeration: The hacker will request the system to divulge the extension numbers. Once he gets this info, he can then start looking for the passwords. The STM will not give out this info.
  3. Password Cracking Attempt: The hacker will try different user names and passwords in order to gain access to an extension or the admin panel of the PBX. The STM can be configured to block an IP if more than 10 trials are done within 10 minutes, for example.
  4. PHREAKERs: These guys take advantage of your negligence and steal from you without really hacking anything… They just check the most common/default user names and passwords used and if they get lucky, it’s a bad day for the victim.
  5. The Hardcore Scammer: Using scripts and special tools, these criminals know exactly what they are doing and have the knowledge to hack and exploit an unprotected phone system. The list of scams they can run is long but it can range from setting up an extension in your system and using it to sell cheap international calls, to more elaborate FAX back or CALL back scams where they use your system to call very expensive / minute phone numbers they control…
  6. DoS/DDoS attacks: These are designed to flood your PBX with an exaggerated numbers of packets. Their goal is to bring down your communication system and render it unusable. The STM will dynamically block for a pre-determined period of time, the IP or IPs from which these attacks originate.
  7. Cross Site Scripting attacks: These are amongst the most complex and hard to achieve. A script is injected in your PBX by the hacker and can program it to do all kind of malicious actions such as having all your extensions ring at once. The STM blocks off the intent and IP address (es) trying to do that.

Manufacturer’s message: The ALLO.com STM uses the real-time deep packet inspection engine, which is in fact a large database of known threats to PBXs. Much like a terrorist watch list, the STM uses this list to check each SIP packet heading towards your system and blocks any malicious packet as well as its originating IP.

Instead of losing thousands of dollars due to the victim of VOIP attacks, invest on 300$ worth of ALLO STM, which is plug & play.

Investing in an STM to protect your communications network is a must.

For more info, visit: http://shop.voipdistri.com/VoIP-PBX-Hardware/VoIP-PBX/Security-Devices/ALLO-SIP-Threat-Manager–aSTM—Up-to-50-Concurrent-calls–SIP-Security-Device–Analyze-SIP-packets–SIP-Protocol-Anomaly-detection.html

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