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VOIP PBX and Servers

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Please list information about VoIP PBX and Servers on this page. Please keep VoIP PBX and server provider information in alphabetical order, and below any other relevant information.

Numeric

  • 1comms.co.uk:Asterisk-based converged telephone system for UK Businesses
  • 1GATE VoIP PBX by Wangate: Cheap VoIP PBX with hardware DSP. Optional internal gateways for Analog/ISDN/PRI/GSM. VoIP resellers welcome.
  • 2N NETSTAR PBX, virtual PBX: VoIP PBX system
  • 2N Omega IP PBX: VoIP PBX system
  • 2N VoiceBlue Enterprise: Simple VoIP SIP PBX
  • 3CX: Windows IP PBX / VOIP Phone system
  • 4PSA VoipNow: Hosted PBX software for service providers and enterprises, accelerating SaaS deployment. It runs on Linux environments (RHEL, SuSE Linux, CentOS, Fedora) on x86 and Power PC architecture based servers.
  • 8ix Zenith: 8ix Zenith spells an Asterisk derived IP Telephony application with the most advanced calling and communication features.


A

  • ActivePBX™ | Turn-Key Business Phone System $149/mo.
  • Aculab Cloud is a framework that enables developers to add telephony to their applications
  • Aksys Networks offers the KONNECT Office Phone System which unlike phone systems requires no PBX, server or Hosted service. The phones create an adhoc network that shares phone services in an office or seamlessly across multiple sites. ...

GSM VoIP Gateway Series from HyberTone

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1, 4, 8, 16 ports VoIP GSM Gateways for call termination, HyberTone Technology

Models of VoIP GSM Gateways
Series 1 SIM port 4 SIM ports 8 SIM ports 16 SIM ports
External AntennaGoIP-1 GoIP-4 GoIP-8 GoIP-16
Internal AntennaGoIP-1i GoIP-4i



1, 4, 8, 16 ports GSM VoIP Gateway ( External Antenna series )

GoIP-1_300.jpg
GoIP-4_300.jpg
GoIP-8_300.jpg
GoIP-16_300.jpg


1, 4 ports GSM VoIP Gateway ( Internal Antenna series )

GoIP-1i_300.jpg
GoIP-4i_300.jpg



Specification


Basic Features:

    • For call termination (VoIP to GSM) and origination (GSM to VoIP) }
    • Standard SIP & H. ...

VOIP Service Providers Business Europe

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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral labanguage and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...

Hosted PBX

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Hosted PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

"Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX


There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

More Hosted PBX Service Providers



1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • Actual US CLEC
  • Branded customer portals
  • Multiple locations on one virtual PBX
  • Customized pricing.


AstraQom Canada | AstraQom Canada is an innovative provider of Hosted PBX, VoIP IP Telephony, Hosting, and Web Services for Business.
  • All Packages Customizable
  • White label/turnkey packages available
  • Virtual Numbers, Vanity Numbers and Toll Free Numbers
  • AstraQom can provide unified communications for nearly all levels of business


.e4 Technologies - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.

  • 1 MONTH FREE TRIAL
  • VoIP and PSTN Service
  • PBX and ACD
  • Web, Audio & Video Conferencing
  • Skills Based Routing
  • Private/Encrypted and Public IM
  • Voice and Data Archiving
  • Reporting and Archiving
  • PSTN Gateway
  • CRM Integration
  • Web Contact Center



1aVOIP.com

1aVOIP.com delivers hosted PBX services with SIP& Mail feature sets. ...

Cloud PBX

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A cloud PBX system is a based on cloud computing technology, where data is stored and transferred over the Internet, rather than on a computer or piece of hardware that an end-user owns.

Cloud technology that has been around for many years, but has only more recently become popular for consumer use with the introduction of programs like MobileMe and the iCloud, both consumer-based cloud computing technologies.

History of the Cloud

Cloud technology has many applications for businesses and consumers. In the mid-1990’s, companies that had previously used “point-to-point” data circuits to move and store data began to use a cloud technology instead as they found that this was a cheaper way to do the same tasks, and made more efficient use of bandwidth.

The term “cloud” likely refers to the drawings of clouds used by its program developers to represent the abstraction of the Internet. The symbol was used to mark the division between responsibilities of the service provider company and the responsibilities of the customer.

Cloud technology means that you can access the data and services of your cloud from any device connected to the Internet. So, with a cloud PBX provider, you can use all of the standard PBX features without the need of an actual PBX in your home or office.

What is PBX? What is Cloud PBX?

PBX, or public branch exchange, is the name used to refer to the technology that any given telephone provider uses to route calls. Originally, a PBX was a large unit of hardware that had to be stored on site and operated by hand. These machines could be very costly for a company, as they were large, needed special storage facilities, needed regular repairs, and had to be operated manually.

Companies with multiple phones and many employees can’t function properly without some kind of PBX technology in the office. In more recent years, a PBX could be made to operate automatically, without an operator, but even these devices were expensive and complicated (the cheapest are around $400). And if a business needed to move to a new location, or add or remove a phone line, the process could be very costly and time consuming.

Initial cost and upkeep of traditional PBX vs cloud PBX

  • Traditional PBX: About $5,000 including cost of installation and new equipment fees. $10.00/hour for an operator. If managed without an operator, can still run to around $300 or $400 for upkeep and Internet connection.
  • RingCentral Virtual PBX: $19.99/month plus $0.049/additional minute
  • Vocalocity Virtual Extension:$14.99/month plus $0.03/additional minute
  • Switchvox Cloud by Digium: $29/month, unlimited minutes and all UC features included

Cloud PBX gives you access to PBX capabilities without the need for complicated and expensive equipment. Cloud PBX is a kind of hosted PBX or virtual PBX, meaning that all of the routing services are performed by your VoIP provider, rather than by an employee managing a PBX system in your office.

How does it work?

All cloud technologies are dependent on the Internet. A cloud PBX works by connecting to your IP, or Internet phone, for its internet connection. So, you must have a VoIP or other IP system in place in your office in order to use a cloud PBX. Many VoIP service providers, like the following, will include their PBX options at little or no extra charge:
  • RingCentral (included in some plans)
  • MondoTalk (included in all plans)
  • Vocalocity (available as a free feature on all plans)
  • Jive Communications (available as a free feature)
  • M5 Networks (included)

You can access your cloud PBX with a personalized account from any device that can run the Cloud PBX’s interface software. This means that you can access your cloud PBX from any Internet-enabled location or device that you have certified for use with your system. Usually, this is just a matter of logging in with a password or security question from that device. ...

VoIP Gateways

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Information about VoIP gateways, including VoIP media gateways, FXO gateways, and other VoIP gateways can be found on this page. If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

Media Gateways

Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone networks and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.

Analog FXO gateways


WebRTC

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Synopsis

The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.

Overview

customLogo.gif.png

WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.

Architecture

The architecture of WebRTC, as described by W3C looks something like this:
WebRTCpublicdiagramforwebsite (2).png


Design

Major components of WebRTC include:

  • getUserMedia, which allows a web browser to access the camera and microphone
  • PeerConnection, which sets up the audio/video calls
  • DataChannels, which allow browsers to share data via peer-to-peer

Support

Chrome WebRTC Development Team

Discussion List: https://groups.google.com/group/discuss-webrtc
Google Plus Page: https://plus.google.com/113817074606039822053
Chrome WebRTC Issue Tracker: http://code.google. ...

SIM Bank / SIM Server from HyberTone

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SIM Bank/ SIM Sever


SIM Bank allows SIM card safekeeping, centralized management. It connected with Hybertone GSM VoIP Gateway via internet.

SIM cards no longer need to be installed in GSM VoIP Gateway anymore. You can deploy your VoIP GSM Gateway in different locations. Centralize and supervise all SIMs in one place, manage the entire system from your office.


SIM Bank_300.jpg


Specification


Major Advantages:

    • Lightweight and Portable
    • Save on travel expense and time
    • Save the cost of 'on site' technicians
    • Save the down time between maintenance
    • Control and manage this system from the office via IP

Function of SIM Server:

    • Rotate SIM Cards on Duty
    • Set GSM Group (Assign several SIMs Per GSM Port)
    • Set Talk Time per SIM, Set Day of week, Set Time Range
    • Change IMEI
    • Monitor CDR, ASR, ACD

Hardware characteristics


    • 32 SIM Card Slots
    • 1 PC port & 1 LAN port (10/100 Base-T)
    • LED Indicators: Power, Status and errors indicator
    • Reset button for device reboot / factory reset

Software characteristics

    • Operating system: Linux V2.6
    • SIM Card standard: GSM11.11 3.3V
    • IP standard: IP V4
    • LAN Port Configuration: DHCP, PPPoE, Static IP
    • Preferred Network Environment: Packet Loss rate <1%, Packet Delay < 300ms
    • One-way SIM peak traffic: 11KBPS

Physical Properties


    • Color: Grey
    • Case Material: Plastic
    • Weight: 1.5KG (Including AC/DC Adapter)
    • Size: 150mm (W) * 290mm (L) * 50mm (H)
    • Power: Input:100-240V~50/60HZ 0.6A Max Output:12V-2A
    • Power consumption: 10W
    • Environment Temperature: (Operating) 10 ℃~ 45 ℃, (Storage) 0 ℃~ 5 0℃
    • Operating humidity: <90%, non-condensing
    • Warranty:One year



Lead Sales: Leo

website: www.hybertone.com

telephone: +86 15118089575

skype: leohybertone

MSN: leo@hybertone.com

yahoo: leovoip007@yahoo.com


Facebook
http://www.facebook.com/HyberTone.VoIP

Youtube
http://www.youtube.com/HyberTone



Asterisk Consultants Greece

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Asterisk consultants: Greece


Modulus SA

VoIP Service Provider
Virtual PBX, SIP Trunking, Greek DID numbers, VoIP Termination, Custom application development that requires VoIP technology, Asterisk PBX


Netvoice.gr

VOIP solutions & IT Services

VOIPsystems.gr

Advanced VOIP telecommunication solutions based on Asterisk.

MerIT Advanced IT Solutions, Athens

Voip Telephony, Servers & Corporate Networks, High Performance Computing.

Transcom Ltd., Athens

Voip Telephony, Asterisk Intergrators

SEQUENCE ,Athens


Application Service Provider, Telephony Software development, System Integrators, Networking Solutions, Asterisk Custom Applications.
Products:QUATTRO Provider -Contact Center- Corporate | CONCEPT Ivr - UM -Sip Proxy | Asterisk CTI QUATTRO |SMS INTERACTIVE GATEWAY

Aptus FonB

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The Phonebook Solution For Asterisk – Aptus FonB is a software product to be released soon by Aptus Telecom, developed to provide integrated contacts from Highrise, Google and Mobile right on your IP Phone registered with the Asterisk Telephony Engine. It makes easier for a user to merge and sync contacts so that a unified phonebook is provided in IP Phone. FonB features contacts from different platforms (Google, Highrise, Mobile) along with the comprehensive Call History. It also allows seamless Call Switching between multiple devices. FonB is expected to work with major Asterisk distros like FreePBX, Elastix etc.

Brief details of FonB features are listed as under.


1. Contacts

Aptus presents 'Contacts' as the most unique and strongest feature of FonB. The idea of FonB basically revolves around syncing multiple contact lists from different devices in one unified directory. In technical terms, sophisticated use of XML flow and event triggering binds all the main constituents of the system ('Google Contacts', 'Highrise CRM' and Asterisk). Security tokens by Google and Highrise CRM also play a vital role in the efficient handling of contacts.


1.1. Contacts Synchronization

This feature allows users to synchronize their contact lists across multiple devices. ...

Wildix

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LogoWildix_2011_blu.png


Wildix


Wildix is a multinational company operating in the branch of telecommunications and specializing in Voice Over IP (VoIP) products and Unified Communication solutions. It operates in Italy and France and uses its own research and development center in Odessa. Bringing the people closer and improving the production at the workplace are the challenges that the United Communications intend to take up, for the world to become always closer and for the work to be always simplified by the new technologies.




WMS— Wildix Multimedia system developed by Wildix



WMS20 (1).jpg



Wildix systems are based on TLC industrial electronics with microprocessor for management and processing of voice and data network protocols. The switch matrix of the system is realized on most software layers of the PBX physical memory. Wildix systems allow convergence of voice and data and they were designed for both SIP and TDM protocols management. Wildix management system is completely web-based and consists of WMS and WMP.

Both components do not require any physical installation and are accessible from PC or mobile devices on any operating system (Windows, Linux, Apple MacOS). Wildix management system can be hosted on the dedicated servers and allows access in a multiuser mode. Via the graphical interface you can activate and manage all your Wildix PBXs (it’s possible to connect all your offices to the WMS Network), add the users and edit their preferences, configure all the devices belonging to your network, even remote ones, thanks to the auto-provisioning system.

Main Features


  • IAX2 SIP users and trunks full
  • automatic autoprovisioning for W-AIR, Cat-IQ cordless phones
  • features presence monitoring (DND, CF, TIMETABLES)
  • inter system presence monitoring and IM system configuration (NETWORKING, DHCP SERVER, NTP SERVER)
  • full extensions management
  • presence monitoring of sip trunks
  • backup and restore of all system properties
  • WMS users ACL remote fs support and usb devices for data storage
  • remote cdr configuration and Blues
  • XML application support




Wildix VoIP Phones



phones.jpg


Phone is an essential instrument we use in our daily work life. The line of Wildix VoIP phones is an ideal complement to your Wildix PBX.
Wildix offers four phone models, each of them was designed to guarantee the maximum audio quality, high performance and usability.
WP480 and WP490 have a clear and intuitive user interface. WP600 is provided with a high resolution TFT color display and HD wideband audio.
W-AIR phone is a Wildix cordless solution that uses CAT-iq protocol for transferring data and voice on the radio channel.

In addition to the basic functions, Wildix phones are supplied with the advanced features, such as XML browser and PoE alimentation. Thanks to the Wildix auto-provisioning system, configuration of Wildix VoIP phones is immediate and simple. You can set up the BLF keys of your phones to monitor different functions of the PBX. For example, via the BLF keys it’s possible to check the activity status of your colleagues and pick up a ringing call of another user, to monitor timetables. ...

Asterisk func message

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Synopsis:

MESSAGE(argument)
new in Asterisk 10



Description:

MESSAGE(argument)
  • argument can be:
    • to - Read-only. The destination of the message
    • from - Read-only. The source of the message. If you want sent message from Asterisk and need change from, use Asterisk cmd MessageSend
    • body - Read/Write. The text of message.


Example


Send message from Asterisk.

exten => 123,1,Set(MESSAGE(from)=sip:11111) ; no affect, no error, from is read only
exten => 123,n,Set(MESSAGE(body)=test message) ; don't use "
exten => 123,n,MessageSend(sip:22222,sip:11111)

See also

Net2Phone Reviews

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Reviews for the Business VoIP and SIP provider Net2Phone

Net2Phone, is the largest distributor-based VoIP provider in the world. With a global distribution network serving foreign-born communities, Net2Phone provides industry leading voice termination rates. Net2Phone offers a full suite of high-quality, low-cost, VoIP solutions to residential, corporate, and mobile customers in over 160 countries worldwide.

Net2Phone SIP Trunking is a cloud- based phone solution in which we deliver high quality international calling by bundling a VoIP SIP Trunk into existing IP-PBXs and/or VoIP supported gateways. This enables enterprises to replace their legacy phone switches and merge their communications via a single IP circuit, saving considerable infrastructure costs for businesses.
Net2Phone’s SIP Trunking solution will enhance communication, enrich collaboration, and provide measurable cost efficiencies that will improve the way you do business.

In addition to significant cost savings and leading international rates, Net2Phone’s SIP Trunking solution involves no additional hardware or operational costs. There is no charge per channel and simple pay-as-you-go pricing. With certifications by Avaya, Digium, Elastix, and other PBX providers, Net2Phone’s SIP Trunking is compatible and recommended for use with most IP-enabled PBXs and Gateways.

Additional Features include:
  • “Follow Me” VoIP Anywhere capability
  • Automatic recharging
  • International DIDs (international virtual numbers)
  • Gateways Available for Legacy PBX’s
  • Support for all major gateways (AudioCodes, Quintum Cisco ATA,SNOM, Grandstream, ATCOM, Elastix)
  • Support for both digest and IP authentication registrations
  • Support for both high quality and low bandwidth codecs (G.711, G.729, G.723)

Write a Review


Foehn Reviews

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Foehn is a business hosted VoIP provider in the UK. Below are reviews about Foehn.

We are IP Communications experts. We offer more than great value hosted VoIP telephone systems.

Founded in 2000 by Technical Director James Passingham, Foehn challenges the status quo of VoIP Communications, thinking differently to help your business become more agile, reduce your costs and become more competitive.

Through our experts in open source technology and networks, we can provide tailored, scalable business IP communications services. We have a track record of deploying only those cutting-edge technologies with proven extensibility, agility and cost effectiveness.

Why choose Foehn?

• We are the open source communications experts
• We are IP communications experts in transitioning to new technology from legacy systems
• We have optimised mobile content networks for the top MNOs in Europe
• We offer expert design, deployment and support

We offer IP Telephony, Contact Centres, Network Services and Data Networking.

At Foehn, we have the expertise to design, develop and support your IP communications to the highest standard.

Write a Review


Telappliant Reviews

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Telappliant is a hosted business VoIP provider in the UK.

  • Official UK Distributor for Digium, Grandstream, IPDialog, Swissvoice
  • Channel Partner for Eicon Networks, Jasomi, Mediatrix, Vegastream, Sipura, SNOM, Speedtouch
  • Internet Telephony Solutions Provider (IP PBX, Bespoke solutions)
  • VoIPtalk Connect - VoIP Termination Service, featuring incoming numbers from over 600 UK cities, International numbering from Austria, Belgium, France, Germany, Italy, Netherlands, Spain, USA. Low outgoing call rates. White label solutions for service providers. Easy online topup and purchasing.
  • VoIPtalk Direct - online shopping site for most well known products. Price match / beat guarantee.
  • Protocol: SIP, IAX
  • UK-based

Write a Review



MOR

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Class 4/5 Softswitch with Billing and Routing



Image


www.kolmisoft.com

MOR is class 4/5 softswitch with billing and routing with extended functionality, increased stability and professional support directly from the developers.




Image


MOR covers such Business Models:


  • Wholesale
  • Retail
  • Calling Cards
  • Call Shops
  • Callback
  • Batch dialer (Auto Dialer)
  • Various other (minor)


Image

snom 720

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Snom 720 4-Line SIP based IP phone


snom720_anthgrey_high_right_Idle_WEB_01.jpg


The snom 720 phone addresses office users that require excellent audio and a large number of programmable PBX-style keys. It combines a state-of-the-art hardware with the proven snom SIP software.
All in all, the snom 720 raises the bar for VoIP phones in its class both in terms of voice quality, available features and day-to-day usability.

The new audio subsystem was designed for excellent performance in office environments. It supports a large range of codecs, wideband audio, handset and hands-free operation according to TIA-920 standard. The specially designed grips which hold and release the receiver compliment overall user-friendliness by eliminating the noise normally associated with lifting and replacing the receiver.
The new foot stand allows two positions for the device (33° and 55° angle). It can be setup and changed easily without tools. The snom 720 comes in a dark grey color that fits the typical office dress code and makes usage traces less obvious than lighter colors.


Highlights

  • 4-line B/W display
  • 18 LED function keys
  • 12 identities
  • Wideband audio
  • Hands-free operation
  • Power over Ethernet (PoE)
  • WLAN / Bluetooth Headset ready
  • VLAN
  • 2 x Gigabit LAN ports

General Features

  • Weight: approx. 900g (670g without foot stand)
  • Dimensions: approx. (mm) 238x190x40 (LxWxD) w/o footstand, 238x190x220 (LxWxD) with footstand
  • Certifications: FCC Class B, CE Mark
  • Safety: IEC 60950-1:2007
  • Average power consumption with PoE: Approx. 2.0–2.3 W (without USB peripherals)
  • Power over Ethernet/PoE: IEEE 802.3af, Class 2
  • Power: Input voltage 5V DC, max. 5 W (adapter not included) (1)
  • Ethernet: 2 x IEEE 802.3/1 Gigabit switch
  • 1 x LAN, 1 x PC: RJ 45
  • USB port: 1 x type A, USB 2. ...

Inoteska

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Inoteska
Inoteska is a Slovak producer of effective solutions for voice&data communication. For more than 20 years, Inoteska has been providing different telecommunication equipment for mobile, fixed line, alternative carriers, private networks, internet service providers, small&medium enterprises and others from all over the world.

Use Inoteska products for inovation, optimalization and expansion of your network.Reduction of costs for calls and data transfer. Optimal use of your existing network and lines! Inoteska provides you the efficient turn-key solution according to your individual needs!

Product range includes:

1.VoIP GSM Gateway/VoIP PBX : VOICE & DATA ROUTER BA

Device supports :

1 E1, 4/8 BRI up to 32 FXS, up to 16 FXO, up to 6 E&M, up to 12 GSM quad-band 850/900/1800/1900Mhz
2 ETH LAN,WAN
1 RS 232 Console
1 USB mini OTG
Power supply 230V AC / -48V DC 50/60 Mhz




2.VoIP GSM Gateway/VoIP PBX : VOICE & DATA ROUTER BB

Device supports :

1 E1, 8/12 BRI up to 32 FXS, up to 16 FXO, up to 6 E&M, up to 12 GSM quad-band 850/900/1800/1900Mhz
2 ETH LAN,WAN
1 RS 232 Console
1 USB mini OTG
Power supply 230V AC / -48V DC 50/60 Mhz


3.VoIP GSM Gateway/VoIP PBX : VOICE & DATA ROUTER

Device supports :

4 E1, 4 BRI up to 32 FXS, up to 16 FXO, up to 6 E&M, up to 12 GSM quad-band 850/900/1800/1900Mhz
2 ETH LAN,WAN
1 RS 232 Console
1 USB mini OTG
Power supply 230V AC / -48V DC 50/60 Mhz





4. Universal Astfin/Asterisk VoIP board : VoIP-PRI/BRI/FXS,FXO,E&M/GSM

Device supports :

1 E1, 4 BRI up to 32 FXS, up to 16 FXO, up to 6 E&M, up to 12 GSM quad-band 850/900/1800/1900Mhz
2 ETH LAN,WAN
1 RS 232 Console
1 USB mini OTG
Power supply 230V AC / -48V DC 50/60 Mhz



5. Asterisk PCI cards BRI - ASTERISK PCI CARD FOR PC BRI 4S0

6. ...

snom 760

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Snom 760 4-Line SIP based IP phone


snom720_anthgrey_high_right_Idle_WEB_01.jpg


The snom 760 phone addresses office users that require excellent audio, PBX-style keys, and rich visual information. It combines a state-of-the-art
hardware with the proven snom SIP software.
The snom 760 not only provides you with comprehensive IP telephone functionality, but also a whole range of extra features which really put it in a class of its own compared to similar products.

The new audio subsystem was designed for excellent performance in office environments. It supports a large range of codecs, wideband audio, handset and hands-free operation according to TIA-920 standard. The specially designed grips which hold and release the receiver compliment overall user-friendliness by eliminating the noise normally associated with lifting and replacing the receiver.
The new foot stand allows two positions for the device (33° and 55° angle). It can be setup and changed easily without tools. The snom 760 comes in a dark grey color that fi ts the typical office dress code and makes usage traces less obvious than lighter colors.

Highlights

  • High-resolution 3.5”color display
  • 12 LED function keys
  • LED function keys
  • Wideband audio
  • Hands-free operation
  • Power over Ethernet (PoE)
  • WLAN ready
  • VLAN
  • 2 x USB (Type A)
  • 2 Gigabit LAN ports

General Features

  • Weight: approx. 920g (690g without foot-stand)
  • Dimensions: approx. (mm) 238x190x40 (LxWxD) w/o footstand, 238x190x220 (LxWxD) with footstand
  • Certifications: FCC Class B, CE Mark
  • Safety: IEC 60950-1:2007
  • Power over Ethernet/PoE: IEEE 802.3af, Klasse 2
  • Power: Input voltage 5V DC, max. 10 W (adapter not included) (1)
  • Average power consumption with PoE: Approx. 2.2–3.4 W (without USB peripherals)
  • Ethernet: 2 x IEEE 802.3/1 Gigabit switch
  • 1 x LAN, 1 x PC: RJ 45
  • USB ports: 2 x type A, USB 2.0 Hi/Full/Low Speed-compatible host interface
  • Handset: 4P4C connector
  • Headset: 4P4C connector or wireless via USB adapter(2) or snom EHS Advanced(3)
  • Color: Anthracite gray


User Interface

  • High-resolution color display
  • 8. ...

Snom Phones

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Snom VoIP Phones and SIP Proxy/PBX


www.snom.com

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Snom SIP based phones:

  • Snom M9: DECT based cordless solution. Sold as base station with handset/charger station and single handset/charger station. Up to eight handsets can register with the base station and the base station can register up to eight SIP accounts. As with other Snoms, the base station can be provisioned via HTTP and, apparently, the individual handset menus can be configured to lock them down somewhat. Supports IPv6, SIPS and SRTP. Does not include an answering machine.

  • Snom M3: To be presented at VON autumn 2007, but sneak previewed at Astricon Phoenix 2007. DECT based (GAP compliant) cordless solution. Sold as base station with handset/charger station and single handset/charger station. Up to eight handsets can register with the base station and the base station can register up to eight SIP accounts. One base station supports a maximum of three simultaneous conversations. As with other Snoms, the base station can be provisioned via HTTP and, apparently, the individual handset menus can be configured to lock them down somewhat. Availability is quoted as 'end October 2007', but knowing Snom, the reality is to be at least the end of November 2007. UPDATE - It's January 2008 and they are now shipping! Does NOT support IPv6, SIPS and SRTP.

  • snom 820: New phone launched at end of 2008.

  • snom 870: Touchscreen phone, first showcase announced for CeBIT 2009 (March 2009) in Hannover. Video demo HERE

  • snom 710: Supporting four SIP identities and blending an elegant utilitarian design for high usability with high performance and affordability, the snom 710 delivers a rich set of everyday business telephony features critical to today’s real-time, information-driven business workforce.

  • snom 720: The snom 720 phone addresses office users that require excellent audio and a large number of programmable PBX-style keys. It combines a state-of-the-art hardware with the proven snom SIP software.

  • snom 760: The snom 760 phone addresses office users that require excellent audio, PBX-style keys, and rich visual information. It combines a state-of-the-art
hardware with the proven snom SIP software.

  • snom 370: This phone was announced at CeBIT 2007 to be available April '07. High-end device, similar to 360, with higher resolution grey-scale display (240x128), a large LED-light for incoming calls and possible VPN support in future firmware versions. The phone has more RAM and CPU power than the 360; no more rubber keys.

  • snom 320: based on the snom360 design but with smaller, two-line, tilt-adjustable LCD display without backlight. Also, the 320 supports conferences (joined at the phone) of only 3 parties (ie, the phone user and two others). The 360 will conference many parties (exact number not known). ...
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