VoIP Distribution
Sangoma S700 IP phone

Sangoma S700 IP phone

Zero Touch installation makes phones easy to provision

Sangoma Technologies Corporation (TSX VENTURE:STC), a leading provider of hardware and software components that enable or enhance IP communications systems for both voice and data, today announced the launch of a new family of IP phones. Designed exclusively for use with FreePBX and PBXact and featuring Sangoma’s Zero Touch technology, which means they connect and operate quickly and easily, right out of the box.

The product line includes the entry-level s300, mid-range s500, and the s700 for the most demanding business environment. Each phone in the three model series features Power over Ethernet (PoE), so no power cable or outlets are required. They have full duplex speakerphones, dual Ethernet Ports, multi-way conference calling, high definition voice, and have built-in Virtual Private Network (VPN) capabilities.

“VoIP telephones can be complex to install, and manually configuring many different parameters with dozens or hundreds of extensions can take hours,” says Bill Wignall, President and CEO of Sangoma. “With Sangoma’s unique Zero Touch installation there is no complexity, since phones connect automatically with Sangoma’s PBX solutions and can be used as soon as they are plugged into the network. Zero touch provisioning saves customers significant time, frustration and money.”

When customers buy and install any of these Sangoma phones, the redirection server automatically sends the phone line to the Sangoma Private Branch eXchange (PBX) for configuration. Other vendors have redirection servers, but they have to be pre-programmed with the IP PBX.

Sangoma’s PBX solutions are available in two versions. FreePBX, an open source variant and as the PBXact brand, a commercial version. Both are available as software or can be pre-installed on telecom appliances. Only Sangoma can provide Zero Touch provisioning with FreePBX and PBXact.

Each phone also comes integrated with Sangoma EndPoint Manager Software. The software enables users to control global settings, program phone keys, map extensions to MAC addresses and templates, upload images, download new firmware, and scan the network for non-provisioned phones. Sangoma phones also feature Plug and Play deployment; native VPN for secure connections; phone and user management from the PBX control panel; and soft key functions like call flow management, queue management, presence, etc. Since the VPN client software is included with the phone, remote users can easily and securely access an office-based PBX. There’s no need for a VPN router at the remote or home office location. The system also traverses firewalls automatically.

Full Integration with Phone Apps
Once Sangoma’s phones are installed, users can control advanced features directly from the phone. There’s no need to remember feature codes. There’s also no need to log in to the PBX panel to control the feature set.

User applications include: Call Parking, Follow Me, Do Not Disturb, Conference Rooms, Call Forwarding, Time Conditions, Presence, Queues, Queue Agents, Transfer to Voice Mail, Visual Voice Mail, and Log in/out.

Other model-specific features and prices include:

  • s300, with 2 Session Initiation Protocol (SIP) Accounts, dual Ethernet capability, headset compatible and an MSRP of $89 (USD).
  • s500 with 4 SIP accounts, a 3.5 inch full color display, 28 programmable soft keys, headset compatible, dual gigabit Ethernet capability and an MSRP of $149.
  • s700 featuring 6 SIP Accounts, a 4.3 inch full color display, 45 programmable soft keys, headset compatible, dual gigabit Ethernet capability and an MSRP of $229.

Accessories

Accessories such as headset adapters and power supply units (for networks without PoE) are also available. All Sangoma telephones come with a 1-year warranty and extended warranties are also available.

With this new line of smart VoIP phones, Sangoma can provide all the appliances, software and functionality a business needs for a sophisticated IP and Unified Communications system.

Sangoma is taking orders now for delivery in March 2016. Please contact Sangoma Europe Partner directly: www.voipdistri.com or www.voipdistri.co.uk

About Sangoma Technologies
Sangoma (TSX VENTURES: STC) offers a range of hardware and software for voice and data IP communication systems to enterprises, SMBs, carriers, and OEMs in more than 150 countries. Sangoma’s cost effective, quick to deploy, and easy to manage offerings include FreePBX, Session Border Controllers (SBC); the market-leading Express for Skype for Business; VoIP Gateways; Call Tapping; Call Center Software; and Signaling Gateways. Sangoma also continues to lead the market in VoIP-to-PSTN interface cards. For more information visit www.sangoma.com.

 

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.

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

User Interface

Access to GUI demo: Click Here

Username: admin
Password: admin

 

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!

Tiptel 3030 + KM 30-40 Expansion Module

Tiptel 3030 + KM 30-40 Expansion Module

Tiptel has launched a new Entry SIP Telephony Series for small business. The new Tiptel 3030, 3020 and 3010 IP phones and KM 30-40 Expansion module it’s January 2015 available by the new Value-Added Tiptel Distributor VoiPDistri.com.

The new IP phones tiptel 3010, tiptel 3020 and tiptel 3030 will be sales at first at 2015 by VoIPDistri.com. The IP series comes with professional telephony functions and it’s for small business application area which support all added feature in the business. Additional features are different types of auto provisioning, XML  phone book, LDAP, an open SIP standard, HD voice as well as a 3,5″ 262K colors display, 4 VoIP accounts, 47 keys including 14 programmable keys and  expandable by keypad extension with color LCD display. The new phone it the result of user-friendly user interface, very good acoustics powered by HD Voice.

The IP phones Tiptel 3010, Tiptel 3020 and Tiptel 3030 start at a price below euros 60 € without VAT and can be ordered via Tiptel Distributor VoIPDistri.com or our online shop at shop.voipdistri.com. All models are available from January 2015.

To supplement the IP phone family the extension module KM 30-40 is available offering 30-40 additional function keys. In total you can connect up to three modules with Tiptel 3030 IP phone.

 

Fanvil i20T SIP door phone with RFID Card Support & Allo Nano2 PBX for 8 People Conference / 32 IP and 8 TDM simultaneous calls / 75 IP extensions, Up to 6 Analog extensions / Easy to use GUI / 8 Ports (2FXO-4FXS-2 FXO/FXS) / Fax and Voicemail to Email

Fanvil i20T SIP door phone with RFID Card Support & Allo Nano2 PBX for 8 People Conference / 32 IP and 8 TDM simultaneous calls / 75 IP extensions, Up to 6 Analog extensions / Easy to use GUI / 8 Ports (2FXO-4FXS-2 FXO/FXS) / Fax and Voicemail to Email

Allo, a manufacturer of VOIP and security products and Fanvil, a developer and manufacturer of IP telephones, today announces partnership to provide a fully interoperable end-to-end Intercom VOIP solution.

The partnership guarantees a full interoperability between Allo Nano2PBX and Fanvil IP Door phone i20T, providing resellers and end users with a reliable and full-feature communication system solution. ALLO believes this is just the start of the cooperation; the two parties will actively expand more areas of cooperation in the future. Having Fanvil on board will allows ALLO to give more productivity to end user experience.

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 Fanvil and Allo Distributor of  IP phones, VoIP Gateway,  Analog/Digital Cards, PBX Systems and Security Devices. See more at: shop.voipdistri.com or 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 Allo and Fanvil VoIP Solutions
of VoIPDistri.com – click here!

Written on November 2nd, 2014, VoIP-News Tags: , , , , , , , ,
Mitel (ex. Aastra)  6867i  business SIP phone

Mitel (ex. Aastra) 6867i business SIP phone

Mitel® (Nasdaq:MITL) (TSX:MNW), a global leader in business communications, today launched its 6800i series of IP business phones at the Enterprise Connect conference in Orlando, Florida. Utilizing open standards based SIP protocol, Mitel’s 6800i series phones are designed to offer the broadest interoperability and support both on-premise and hosted services with the best path for businesses migrating to the cloud.

“Mitel phones are all about empowering our customers to make the connections they need to drive and grow their business. Adding the 6800i series to our portfolio further expands customer choice with highly interoperable phones that are designed to work with all of the world’s leading IP call platforms. We deliver enterprise-quality voice and features that make our phones highly effective, supporting our customers in their migration to the cloud,” said Dave Johnson, marketing manager for Business Communications at Mitel.

This new SIP phone series comes from Mitel’s recent merger with Aastra and further expands Mitel’s comprehensive offering of business phones and enterprise voice solutions. The latest phones build on the extensive 6700i series feature set, adding high-resolution color displays and GigE interfaces for clearer image and video support. They further enhance the phones’ crystal clear audio and speakerphone performance and include a native electronic hook switch (EHS) headset port. With lower power consumption, intelligent Power over Ethernet (PoE) Class reporting and smaller packaging that includes 100 percent recycled and biodegradable material, the 6800i series is one of the most environmentally friendly family of SIP phones on the market.

Expandable Business Phones for Power Users

Mitel’s 6800i Series SIP phones provide many of the features that are in demand by today’s power users including GigE Ethernet ports to support implementation on high speed networks, phone customization with programmable soft keys and high quality audio. With remarkable HD wideband audio, an enhanced speakerphone and advanced audio processing, the 6800i series delivers crystal clear hands-free conversations. The series also offers color graphical displays, DHSG/EHS headset support and an extensive array of accessories including expansion modules and a detachable keyboard, all adding to the user experience.

The phones will begin shipping worldwide through Mitel partners starting now. For more information, go to 6800i SIP Phones

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 European VoIP Distributor of IP phones, VoIP Gateway,  Telephony Cards, IP PBX and special network solutions. VoIPDistri.com is Voice over IP strategic partner since from 2003, see more at: http://shop.voipdistri.com or http://www.voipdistri.com

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