VoIP Distribution
OpenVox Asterisk-based X204 Hybrid Card

OpenVox Asterisk-based X204 Hybrid Card

OpenVox Communication Co.,Ltd, a global leading provider of the best cost effective VoIP gateways and Asterisk® solutions, announced today the launch of X204 Series Hybrid Card, one of the most advanced Asterisk-based and easiest-to-use hybrid solution available.

The OpenVox X204 Hybrid Card supports all three types of telephony interfaces, integrates Analog, BRI and E1 on one single card, offers a highly flexible choice for different users. More specifically, it accesses Voice over IP world without interfaces restrictions. For example, the FXO or FXS module accesses 2 analog line via juntion box, the BRI module provides 4 simultaneous voice calls over 2 ISDN BRI line, and the E1/T1/J1 module supports standard telephony and data protocols, including Primary Rate ISDN protocol families for voice, PPP, Cisco, HDLC and Frame Relay data modes. Both line-side and trunk-side interfaces are supported.

X204 Series Card works perfectly with open source Asterisk® and enables users to take full advantages of the high performance and rich features of Asterisk®. It will be ready for worldwide shipping today. For more product details, please visit http://shop.voipdistri.com, or you can simply email us at info@voipdistri.com for any enquiries.

Compared with Similar Products

Written on May 23rd, 2016, VoIP-News Tags: , , , , , , , , ,
iAG804 /808 /840/ 880

iAG804 /808 /840/ 880

OpenVox Communication Co.,Ltd, a global leading provider of the best cost effective VoIP gateways and Asterisk® solutions, has announced today to release a brand new series of analog gateways. With features of small size, exquisite enclosure and high performance, the iAG Analog VoIP Gateway is specially designed for saving space and maximizing cost savings which is ideal for SMBs and SOHOs.

Be different from VoxStack series, the OpenVox iAG series has no moving parts and built to last, it includes 4 models and available in 4 or 8 FXS/FXO ports. It can be configured for different country uses, provides a wide selection of codecs including G.711A,G.711U,G.729,G.722,G.723,iLBC.The OpenVox iAG analog VoIP gateway uses standard SIP protocol, includes all analog gateway features and compatible with leading IMS/NGNNGN platform, IPPBX and SIP servers.

Based on the open source Asterisk, the iAG analog gateway comes with high performance, friendly GUI and unique design, it will be ready for worldwide shipping today. For more product details, please visit http://shop.voipdistri.com, or you can simply email us at info@voipdistri.com for any enquiries.

OpenVox and the OpenVox logo are the registered trademarks of OpenVox Communication Co Ltd. All other trademarks are the property of their respective holders.

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!

Written on April 1st, 2016, VoIP-News Tags: , , , , , , , , , , , ,
OpenVox DGW-L1

OpenVox DGW-L1

OpenVox Communication Co.,Ltd, a global leading provider of the best cost effective VoIP gateways and Asterisk® solutions, has announced today to release a low-end but full-featured T1/E1 gateway, provides the most cost-effective way to connect traditional telephone system to IP networks and integrates VoIP PBX with the PSTN seamlessly.

Aim to maximize the cost savings for operators and SMBs, DGW-L1 has full functions of T1/E1 gateway with competitive price and small size. It supports R2 / SS7 / PRI protocol and be compatible with all kinds of SIP servers. Available in 1 port T1/E1, it backs up 30 concurrent calls and allows easy configuration via Web GUI.

“We have started to design DGW-L1 after we launched our first T1/E1 gateway series last year, we want to supply an alternative to the market. ” said Lin Miao the CEO of OpenVox, “Through repeated testing,  we finally make it based on a low-end solution, we anticipte that this will appeal to many customers who are aware of business expenses and product reliability in the current global VoIP market.”

The new DGW-L1 gateway will be ready for shipping worldwide from January 15th, 2016. Please contact your sales representatives for your partner price or you can simply email me at info@voipdistri.com for any enquiries.

 

Written on January 14th, 2016, VoIP-News Tags: , , ,
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!

OpenVox Distributor VoIPDistri.com Voice over IP DistributionOpenvox is happy working with VoIPdistri.com Voice over IP Distribition as strategic partner and value-added distributor in Europe since from 2009, while helps us gain a lot of new reseller, system integrate partner in voip filed.

VoipDistri has full experience and long history in the field of Analog-, GSM-, ISDN-networks and Voice over IP from 2003. VoIPDistri.com is able to offer you a comprehensive service for all Voice over IP requirements and network solutions.

OpenVox Communication Co.,Ltd is a global leading provider of the most advanced open source Asterisk® telephony hardware and software products. VoxStack Hybrid VoIP gateways are the innovative products from OpenVox. They are the industry FIRST Hot-Swap Asterisk-Based VoIP Gateways. This kind of gateways could help reduce expense of devices and save space and make equipment convenient for operation. They provide full flexibility and maximum asterisk functions on the gateways.

These new generation hybrid gateways come with OpenVox unique modular design that supports up to 5 plug-in modules. You could randomly combine different telephony interfaces including GSM, FXO/FXS, BRI or E1/T1. With current version, it just supports GSM/FXS combination (up to 44GSM channels or 88 FXS ports or any combination of five GSM/FXS gateway modules at maximum), other types of combination (FXO/BRI/E1/T1) will come to use soon.

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!

 

Askozia 4.0 releasedAskozia proud to announce that AskoziaPBX 4.0 is now available! One of the main topics of this release is security, with a focus on two areas: The overall security of AskoziaPBX, to protect installations that are on the internet or directly connected to it. To achieve this, we have included a firewall and an automatic brute force detection. Askozia second goal was to make phone calls more secure. Version 4.0 includes state of the art call encryption to keep your phone calls private.

As usual, all new security features don’t require any expert knowledge and just a handful of clicks to be fully functional.

Finally, Askozia like to thank you. Many of the new features and improvements are based on feedback from you. Thank you for helping us making better software!

    • Built-in Firewall
      Maximum protection of AskoziaPBX installations with a built-in firewall and intelligent brute force detection with Fail2ban.
    • Encrypted Phone Calls
      Encrypted communication with SecureSIP and SRTP in an amazingly simple way. Secure Calling is supported for all officially supported desk phones.
    • Dashboard and Improved Usability
      One dozen new features, a polished web interface and a dashboard with direct access to all important system components.
    • New Call Flow Editor Modules
      A calendar module and an enhanced playback module with external access complete the most popular of our software addons.
    • Widened hardware and VM support
      A brand new Linux kernel for a massively increased hardware support and improved performance. Enhanced support for Hyper-V and VMware virtualization environments.
    • Live USB Image
      A USB bootable firmware image with every PC/Server download for easy installation on dedicated hardware without optical drives.
    • Go 4.0 today!
      Unlimited users and voice channelsStarting from
      €149.00 + VAT, if applicable

 

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 Askozia Distributor. VoIPDistri.com is Askozia strategic partner since from 2010, see more at: http://shop.voipdistri.com or http://www.voipdistri.com 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 Askozia VoIP phones of VoIPDistri.com product Portfolio – click here!

 

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

Download

  • Datasheet – Download Now
  • User Manual – Download Now
  • Quick Installation Guide – Download Now

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OpenVox UCP (Unified Communication Platform)

OpenVox UCP (Unified Communication Platform)

OpenVox Communication Co., Ltd, a global leading provider of the best cost effective open source Asterisk® telephony hardware and software products, today announces innovative and affordable UCP (Unified Communication Platform) solutions for enterprises to help them to build up their telecommunication systems with a variety of options in a simple way.  The unique modular design of OpenVox UCP enables the solutions are simple to deploy, highly scalable, feature-rich and for the best value.

The OpenVox UCP is more than just a VoIP gateway or an IPPBX. It is a 2U hybrid unified communication platform with both VoIP gateway and IPPBX functions. You can simply customize your IPPBX and VoIP Gateway solutions in one box with different interface modules. The UCP supports FXS, GSM and E1 interface modules and CPU board modules. The CPU board module comes with X86 architecture and works as an independent IPPBX system, supporting both Linux and Windows. The UCP supports two independent IPPBX systems in one box. Thus one system can be set as a master IPPBX system and the other as a redundant system.  All data storage can be saved synchronously in the two systems. The UCP brings you the real all-in-one solution.

“The UCP is a cool platform to integrate hybrid telecommunication solution in one chassis. When we are talking about the UCP, we mean we are simplifying the installation steps and providing the best user experience possible.” said Lin Miao, the president of OpenVox, “The UCP will provide the enterprises of all sizes with feature-rich asterisk-based VoIP gateway and IPPBX solutions, giving customers the best in class open source technology.”

All the hardware products are fully backed up with a 3-Month “No Questions Asked” Return Policy to ensure satisfaction guarantee for all our customers. The UCP will be ready for worldwide shipping today. Please contact your sales representatives for your partner price or email us at info@voipdistri.com for any enquiries.

Family Products

Chassis and Ethernet switch adapter

GSM VoIP adapter

FXS VoIP Adapter

T1/E1/J1 Interface adapter*1

Processor adapter

*1: Interface adapter which is PCI-Express based must be used with processor adapter together.
*2: This adapter occupies 2 slots.

Download: VS-CHS-2120 Datasheet


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 of VoxStax Unified Communication Platform (UCP), VoIP Gateway Telephony Cards, IPPBX and Failover. VoIPDistri.com is OpenVox strategic partner since from 2008, see more at: http://shop.voipdistri.com or http://www.voipdistri.com

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 VoIP Solutions of VoIPDistri.com product Portfolio – click here!

OpenVox IX210 Unified Communication Server is an upgrade version of IX132, an open source asterisk-based complete IPPBX solution for SMB. With affordable price, users may easily setup their customized IPPBX.

OpenVox IX210 Unified Communication Server is an upgrade version of IX132, an open source asterisk-based complete IPPBX solution for SMB. With affordable price, users may easily setup their customized IPPBX.

OpenVox Communication Co., Ltd, a global leading provider of the best cost effective open source Asterisk® telephony hardware and software products, today announces the release of the first 2U IPPBX IX210 to enable more flexibility in hardware expansion. The IX210 equipped with a dual core 1.86GHz Intel processor that can support up to 300 sip extensions and 120 concurrent calls. What’s more, the IX210 comes with a dark blue front panel that simply makes it unique in color.

IX210 supports integrating up to 2 pieces of OpenVox PCI-E telephony Interface cards, with any combinations of analog, BRI, PRI, GSM or Transcoding interfaces. The LCD in the front panel is a plus for the new design. The optional hardware RAID 1 support provides better data backup for your system. The optional HDD case provides the possibility for hot plug function to the storage device. When RAID1 is installed, thus users can see the status of the storage devices from a GUI and then simply replace a backup HDD whenever needed without opening the appliance. Redundant power supply is available for IX210.

All OpenVox hardware products are fully backed up with a 3-Month “No Questions Asked” Return Policy to ensure satisfaction guarantee for all our customers. The IX210 will be ready for worldwide shipping today. Please contact the main OpenVox Distributor VoIPDistri.com for your partner price or you can simply email us at info@voipdistri.com for any enquiries.

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 of VoxStax VoIP Gateway,  Telephony Cards, IPPBX and Failover. VoIPDistri.com is OpenVox strategic partner since from 2008, see more at: http://shop.voipdistri.com or http://www.voipdistri.com

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 VoIP Solutions of VoIPDistri.com product Portfolio – click here!

 

 

OpenVox VoxStax GSM Gateway: The industry 1st opensource based GSM VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design users may easily setup their customized Gateway. Also decondary development can be completed through AMI.

OpenVox VoxStax GSM Gateway: The industry 1st opensource based GSM VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design users may easily setup their customized Gateway. Also decondary development can be completed through AMI.

The VoxStack VoIP GSM Gateway OpenVox GW2120-44G with 44 GSM Channels and VoIP Analog Gateway OpenVox GW2120-88S with 88 FXS Analog Ports can direct buy to webshop shop.voipdistri.com the Leading Distributor of VoIP Telephony! The new OpenVox Gateway family  VS-GW1600  it’s with up to 5 and VS-GW2120 it’s with up to 11 different telephony interfaces including GSM, FXO/FXS, BRI, E1/T1 it’s now on  VoIPDistri.com stock available.

On-line Demo

OpenVox VoxStack GW2120 Series

OpenVox VoxStack Series GSM Gateway is an industry 1st open source asterisk-based GSM VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI.

There are three GSM Gateway models with VoxStack series GSM Gateway, the VS-GW1202-4G, VS-GW1202-8G, VS-GW1600 and  VS-GW2120 GSM series. The Modular Design GSM Gateways are ranging from 4 up to 44 GSM channels, developed for interconnecting a wide selection of codecs, including G.711A, G.711U, G.729, G.722, G.723, G.726 & GSM , to the GSM cellular networks to quickly reduce telecommunication expenses and maximize cost-savings. With the unique design of the VoxStack Gateway, it can support hot-swap for both SIM cards and GSM gateway modules. Users can simply add or remove the modules for hardware expansion or exchange. Each GSM gateway module runs an independent asterisk system inside.

The VoxStack gateway designs with a Lan Switch board that provides stackability on the hardware upgrade. It supports SMS messages sending and receiving and group sending and SMS to email. The GSM Gateways are 100% compatible with asterisk, Elastix, trixbox, 3CX, FreeSWITCH sip server and VOS VoIP operating platform.

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 – Leading  OpenVox Distributor of VoxStax Hybrid VoIP Gateway in Europe and Openvox strategic partner since from 2008, se more at: http://shop.voipdistri.com or http://www.voipdistri.com

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