Connected Cars and safety
To increase safety on our roads, intelligent networking and digitalization inside and outside the vehicle is set to play an increasingly important role in the future. “Connectivity” means that vehicles can communicate both with each other (vehicle- to-vehicle, or V2V) and the road infrastructure (vehicle-to-infrastructure, or V2I) such as stoplights and traffic management systems. This communication – also known under the umbrella term “car-to-X” communication – warns and informs drivers instantly of hazardous situations along the route, even if these are not yet visible to the driver themselves. During highly or fully automated driving, the vehicle would brake autonomously in such cases or change lanes in order to bypass the hazard at a safe distance without the driver having to intervene.
Various communication technologies are available for ensuring the required level of connectivity, including:
- Standardized, general-purpose short-distance technology (BluetoothTM, Wi-Fi, wireless power, NFC etc.)
- Technology developed specially for vehicle connectivity (e.g. IEEE 802.11p, a Wi-Fi-like short-distance communication standard for V2V and V2I)
- Mobile network coverage (GSM, UMTS, LTE and all associated variants)
A comparison of technologies
The enforcement of a ban on using cellphones in the car without a hands-free kit has contributed to the popularity of Bluetooth technology, which allows drivers to control incoming and outgoing calls via the dashboard and connect the audio signal to the hands-free microphone and loudspeaker in the vehicle. Standardization was advantageous here because the Bluetooth Special Interest Group has developed a specific profile for this scenario: The hands-free profile (HFP).
Wi-Fi is the certified and generally preferred process for providing vehicle occupants with infotainment services. The car itself can act as a hotspot. Wireless power allows wireless charging of cellphones, smartphones and other devices without any action on the part of the user – that is, without the driver having to perform any distracting task; at the same time, the mobile device is always ready to communicate (in the car via Bluetooth and fully charged when the driver leaves the vehicle). IEEE 802.11p – a technology similar to Wi-Fi – was developed to facilitate V2V and V2I communication. There is, however, still a long way to go before this technology becomes widespread in the automotive industry because it can be leveraged to its full potential only when used on a mass scale and the necessary investments are made in the (road) infrastructure.
In the field of connectivity, mobile communication technologies are not only an important basis for V2V and V2I communication but also the key to on-board eCall emergency call systems, which, by March 31, 2018, will be mandatory EU-wide in all cars and light-duty commercial vehicles presented for homologation at that time. In the event of a serious accident, the system ensures that emergency medical services are alerted, even if the driver or other vehicle occupants are themselves unable to make an emergency call or speak on the cellphone. According to the European Parliament, eCall could potentially reduce the number of traffic fatalities by 10% per year. The member states are required to install the necessary infrastructure by October 1, 2017.
Ensuring connectivity is a key safety requirement
eCall is standardized for use in 2G (GSM) or 3G (UMTS) networks, but not in 4G (LTE) networks – network operators, however, are already implementing 4G and are currently testing future 5G networks. Although 2G networks have universal coverage in Europe, they are set to be disabled in the not-too-distant future. 3G networks already have good coverage in Europe. Something else that has to be considered, however, is the frequency band. In Europe, there are multiple frequency bands used for 2G and 3G, which means that an eCall modem has to support different frequency bands to ensure interaction with mobile communication networks in the whole of Europe. LTE/4G is a mobile communication network featuring state-of-the-art technology only recently introduced by network operators. However, LTE is a non-voice technology used exclusively for transmitting data.
Most smartphone users want high-speed data transmission, but are unaware that this technology does not support voice calls. Voice calls are possible only because the telephone itself switches down to 3G mode when a call is received or the user makes a call, although this is set to change when the new VoLTE technology is launched, which is currently undergoing testing and which some operators have already introduced. Test programs for these devices should therefore definitely ensure that eCall is supported not only by 2G or 3G cellphones and modules but also by 4G cellphones and modules.
In summary, the functions featured in most “connected car” applications rely on communication technology. For non-safety-related applications, the loss of signal is not critical – users can easily check whether or not connectivity is available. For safety-related services and applications like eCall, however, warnings should be issued to inform users of any loss of communication capability. The system should also be able to automatically restore functionality as soon as the signal is stable again.


A Global Company
GSM-based mobile communication in Turkey began when Turkcell commenced operations in February 1994. Turkcell then signed a 25-year GSM license contract with the Ministry of Transportation on April 27, 1998. Since then, it has continuously increased the variety of its services based on mobile audio and data communication, as well as on its quality levels and as a result, its number of subscribers.
Turkcell is a regional leader by being the market leader in five countries out of nine it operates in. Turkcell's shares have been traded on the Istanbul (IMKB) and New York Stock Exchanges (NYSE) since July 11, 2000, and it is the first and only Turkish company ever to be listed on the NYSE.
Turkcell also has investments abroad. Turkcell conducts its mobile operations in growing markets like Azerbaijan, Kazakhstan, Moldova and Georgia through its subsidiary Fintur, in which Turkcell holds a 41.45% stake. KKTCell, a 100% owned subsidiary of Turkcell, was established in 1999 and operates within the framework of a revenue-sharing agreement with the Telecommunications Authority of the Turkish Republic of Northern Cyprus.
Astelit, a company in Ukraine in which Turkcell holds a 55% stake, began operating in February 2005 under the brand life:). Turkcell acquired 80% stake of BeST from the State Committee on Property of the Republic of Belarus in 2008.
Meanwhile, Turkcell's wholly owned subsidiary in Germany, Turkcell Europe, commenced its operations in 2011. In the first quarter of 2011, Turkcell expanded its operations into nine countries by initiating MVNO services in Germany through the wholesale traffic purchase agreement signed in 2010.
Turkcell, with its wide coverage area and diverse range of services abroad, is able to provide its subscribers with mobile communication services both in Turkey and around the world.
Turkcell had signed 3G contracts in more than 110 countries, ranking it among the world’s top operators in terms of the provision of international data services. Turkcell also ranks among the top operators in terms of GPRS roaming, where it has signed contracts with operators from more than 165 countries.


Railway regulations on electronic devices
At Biosphere Flux we design, project and produce lights, plugs or USB charging ports that comply with international standards for the railway sector EN 50155 and EN 50121-3-2.
EN50155 – The standard that covers electronic equipment for railway applications
The EN50155 certification is an International Standard that covers electronic equipment used on rolling stock for railway applications. The Standard covers aspects of this electronic equipment, including temperature, humidity, shock, vibration, and other parameters.
EN 50121-2-3 – The European standard covering the emission and immunity aspects of electromagnetic compatibility
This European standard applies to the emission and immunity aspects of electromagnetic compatibility of electrical and electronic equipment intended for use in the railway rolling stock.
The objective of this Standard is to define the limits and test methods for electromagnetic emissions and the requirements for immunity tests in relation to conducted and radiated disturbances. These limits and tests represent the essential requirements for electromagnetic compatibility.
The emission requirements have been selected in order to ensure that the disturbances generated by the electronic device operate on railway rolling stock and do not exceed a level that could prevent other devices from operating properly. The immunity requirements have been selected in order to ensure an adequate level of immunity to rolling stock equipment.
This international standard applies to industrial electrical equipment operating in the frequency range from 0 Hz to 400 GHz, as well as to domestic and similar equipment designed to generate or use radioelectric energy locally.
An international standard that applies to industrial, scientific and medical electronic equipment operating in the frequency range from 0 Hz to 400 GHz. Measurements are only necessary in the frequency ranges for which limits are specified in chapter 6.
For Industrial, Scientific, and Medical (ISM) applications of radio frequencies, within the meaning of the definition given in the ITU Radio Regulations, this standard covers the emission requirements related to radio frequency disturbances in the range of frequencies from 9 kHz to 18 GHz.
This standard specifies the requirements for radio frequency ICM lighting equipment and for UV radiation generators operating within frequency bands in the frequency range 9 kHz to 18 GHz.
Equipment covered by other CISPR product family and product emission standards does not fall within the scope of this standard.
This part of the IEC 61000 Standard concerns impunity requirements and test methods for electrical electronic equipment subjected to discharge of static electricity produced directly by the operators and between people and objects located in the vicinity. The ranges of the test levels relative to different environmental and installation conditions are further defined and the test procedures are established.
The purpose of this standard is to establish a common and reproducible basis for evaluating the performance characteristics of electrical and electronic equipment when subjected to electrostatic discharge. In addition, the discharges that operators can produce on objects close to vital equipment are included.
This part of IEC 61000 applies to the immunity of electrical and electronic equipment to radiated electromagnetic energy. Defines the required test levels and procedures.
This section is intended to establish a common reference for the evaluation of the performance characteristics of electrical and electronic equipment subjected to radio frequency electromagnetic fields. The test method documented in this part of IEC 61000 describes a method to assess the immunity of equipment or systems against the defined phenomenon.
As described in IEC Guide 107 this is a basic EMC publication for use by IEC product committees. As stated in Guide 107, IEC product committees are required to determine whether or not to apply this immunity test standard and, if so, are responsible for specifying the appropriate test levels and conformance criteria. . Committee 77 and its subcommittees are prepared to cooperate with product committees in evaluating particular immunity assay values for their products.
This part deals with immunity tests relating to protection against radiofrequency electromagnetic fields from any source.
Particular considerations are devoted to protection against radiofrequency emissions from digital radiotelephones and other radiofrequency emitting devices.
This part defines the test methods to measure the effect of electromagnetic radiation on the equipment of interest. Simulation and measurements of electromagnetic radiation are not accurate enough to quantitatively determine the effects. The defined test methods have been fine-tuned mainly to obtain a good reproducibility of the results for different test installations for a qualitative analysis of the effects.
This Standard is an independent test method. Other test methods cannot be used as a substitute for giving conformance to this standard.
This part of the IEC 61000 Standard deals with the immunity of electrical and electronic equipment to repetitive fast transients. Immunity requirements and test procedures for electrical fast transient bursts are given. Ranges for test levels are also defined and test procedures are established.
The purpose of this standard is to establish a common and reproducible reference for the purpose of evaluating the immunity of electrical and electronic equipment sustained to fast transient bursts at power, signal, control, and ground ports. The test method documented in this part of the IEC 61000 series describes a consistent method for evaluating the immunity of an equipment or system against a defined phenomenon.
This part of the IEC 61000 Standard refers to the immunity requirements, test methods, and range of recommended test levels for equipment against unidirectional shock waves caused by transient lightning-type overvoltages ( atmospheric) and maneuver type. Various test levels are defined relative to different environments and installation conditions. These requirements have been developed and are applicable to electrical and electronic equipment.
The purpose of this standard is to establish a common reference to evaluate the immunity of electronic and electrical equipment, when subjected to shock waves. The test method documented in this part of IEC 61000 describes a method of evaluating the immunity of a piece of equipment or a system against a given phenomenon.
This part of the IEC 6100-4 Standard deals with the requirements related to conducted immunity of electrical and electronic equipment against electromagnetic disturbances caused by intentional radio frequency (RF) transmitters, in the frequency range of 9 kHz to 80 MHz. Equipment that does not even have a conductive wire (such as power cords, signal transmission lines, or ground connections) that can couple the equipment with RF fields is excluded.
The purpose of this Standard is to establish a common reference in order to assess the functional immunity of electrical and electronic equipment when subjected to conducted disturbances induced by radio frequency fields. The test method documented in this part of IEC 61000 describes a consistent method in order to assess the immunity of equipment against a defined phenomenon.


1. Monitoring power usage data
2. Support mobile APP remote control
3. Support for Amazon ALEXA, for Google HOME, voice control after pairing
4. USB fast charging
Material: ABS material
AC frequency: 50-60Hz
Rated current: 10A
Rated voltage: 100V ~ 250V
Wireless protocol standard: 2.4GHz WIFI
Maximum output power: 2000W
USB output: 5V / 3.4A
Standby power: 2W
Operating environment: -10 ℃ -45 ℃
Control distance: Within 30m (WIFI), unlimited (remote)
Product size: 80 * 80 * 60mm/3.15*3.15*2.36\"
1. Due to the different monitor and light effect, the actual color of the item might be slightly different from the color showed on the pictures. Thank you!
2. Please allow 1-3cm measuring deviation due to manual measurement.
1 X Smart Socket
1 X Manual


For bands, orchestras, and singers, sound is the most important consideration for their performance. Their goal is for their concert sound to be the best it can be when they perform for live events. Live events are not like a recording studio where they can control how they sound to a very high degree with sound processing that corrects off-key and other issues. That’s why many performers can never live up to their recording studio sound when playing live.
The goal for venues and companies providing sound production services is for the audience to hear the performance clearly based on the venue’s acoustic properties, audience size, and the scale of the event. That’s the reason that many entertainers bring their own sound engineer and mixing board to events. It’s worth the cost to them to help assure that the concert sound is engineered right for their instruments and voices.
Audio Setup for Concert Sound
Setting up a concert sound system for a performance requires taking into consideration a range of strategic and technical elements when planning the audio components of such a sound system.
Factors to consider for your event to determine your audio setup needs:
- Size and nature of the performance space
- Acoustic properties
- Expected attendance
- Sound ordinance restrictions
- Artist audio requirements and the type of music being performed
- Overall theme of your event (Is it a music-focused festival or is the music intended to be background noise? Etc.)
The concert sound system components are based on coverage requirements and the size and nature of the venue. While the input side of the sound system may likely be the same for different size events, the output side (amplifiers and speaker system will vary greatly by the coverage requirements, size of audience and the acoustic properties of the venue.
For an acoustic performance or for genres where bass and kick drum aren’t key in front of an audience of a couple of hundred people, your audio setup can be as simple as two full-range speakers and a mixer set up. Smaller party bands will often carry this type of audio setup with them to events.
Contrast that with an outdoor or arena concert that involve tens of thousands in the listening crowd. In these situations, you will need as many as 40 or more speakers. With either setup the audio engineer or sound tech will need to tweak the audio for acoustics for better audio quality. An advantage of outdoor venues is that they usually take place in a relatively open area eliminating the need for most acoustic tweaking. However, the lack of reflecting surfaces will require a more powerful sound system to cover the same area.
Concert Sound Involves Understanding the Following:
- Sound System Components
- Sound Board
- Equalization Process
Understanding Concert Sound Begins with Venue Acoustics
Music venues spend a lot of time addressing the production audio needs for their events. Whether it’s an outdoor temporary stage, a nightclub venue, theater, performance center or even a concert hall, a lot of effort and resources are directed at the PA system with the right mixing board, amplifiers, and speakers for the performances to take place there.
What sometimes is lost in all this preparation are the acoustic properties of the performance site. One of the most recent examples of this failure was the opening of U.S. Bank Stadium in 2016. One of the early concert events was a Metallica concert. The fan experience at that concert was disappointing because of the poor acoustics of this brand-new stadium. Millions of additional dollars were spent after it opened to address the acoustical problems with the facility.
This is a perfect example of addressing sound needs without addressing the equally important acoustic requirements for the venue. No matter what the size of music venues or performance areas, you should be looking at creating a great fan experience. Venue acoustics should be a crucial part of planning for concert events.
Understanding How Acoustics Affect Sound is Fundamental to Planning Music Venues Performance Spaces
The fact that acoustics is a branch of physics concerned with the study of sound can be intimidating to those who provide sound for performances at venues. Some basic understanding of the physics involved can help us provide better concert sound for events.
Audio Versus Acoustics
When we refer to audio, we are generally talking about the electronic representation of an acoustic signal as we create, amplify, store and broadcast sound. To hear the audio, the signal must be turned into acoustical waves. Sound, like light, travels in waves thought air or another medium. We “hear” sound when the sound waves enter our ear. It causes the ear drum (tympanic membrane) to start vibrating. The vibration is amplified as it moves through the middle and inner ear. In the inner ear the vibrations are finally transformed into electrical energy that travel via auditory pathways to the parts of the brain involved in what we “hear” as sound.
If we want to understand how sound affects a listener, we need to analyze the three components involved. They include the source of the sound, the pathway to the receiver, and the receiver itself. During concerts, the receiver is most often a human listener in the audience. To change the sound volume or quality means manipulating one of these three elements.
For example, If you wanted to increase the sound volume in music venues, you can increase the gain on the source or you could move the receiver closer to the source. You can also remove things that absorb sound in the pathway to the receiver as well.
Defining the Acoustics of an Entertainment Space
When a speaker, singer or musician produces concert sound, the sound waves are projected outward from the source and get absorbed or reflected by the surfaces they encounter along the way to the receiver. Acoustical properties or the acoustics of an area are the way the sound waves interact with the music venues spaces’ environment.
The Ways in Which Sound Waves Interact with Their Environment
Except for a large empty field, a so-called free field environment, where the only thing that affects the path is distance, other spaces have objects contained in them that affect the path from the source to the receiver. As sound runs into things, it is affected in several ways. Sound waves, like light waves tend to travel in a straight line.
Most things sound waves impact will absorb some amount of the sound. Absorption may involve sound waves passing through the material or may even cause the sound waves to be converted to heat or another form of energy as it’s absorbed. Be aware that the amount of sound wave absorption versus reflection is frequency dependent for a given material. That means that low frequency absorption may be larger or smaller than high frequency sound wave absorption.
When sound waves encounter an object at least some of it reflects. The amount of reflection will be determined by the angle of incidence and the material it impacts. It behaves much like light in that the angle of incidence equals the angle of reflection, much like the way a billiard ball bounces off the side of a pool table. A large, flat surface functions like a mirror for sound waves. In a typical indoor listening environment, we are hearing sounds that have reflected off numerous objects and surfaces,
When sound waves encounter a reflective surface that is uneven, the waves may reflect in many directions at the same time because of all the smaller surfaces that are found on a surface that is not flat and smooth. The waves have many angles of incidence in such a circumstance, generating many angles of reflection.
Sound waves can bend to go around objects. The bending of sound waves is called diffraction. It happened when the wavelength of the sound wave exceeds the dimension of the object. This enables some of the sound waves to avoid being reflected. Since low sound waves have larger wavelengths, the lower the frequency the more likely they will diffract.
Refraction typically does not impact the sound waves generated during concerts or speaking engagements. It refers to sound bending because of the speed change that results from traveling through a different medium or temperature environment.
Indoor Acoustic Issues
The acoustics or acoustic properties of music venues is the behavior of sound waves within them because of the things they encounter in its pathway between the source and the receiver in the ways explained above. Venues need to optimize their acoustics, considering both the properties of the performance space but also the type of audio that will be played there. Space issues include the size and shape of the room, the materials used for both the structure and furnishings, and the size and configuration of the expected audience. The optimal acoustics are different for an acoustic performance as opposed to a rock band performance. Most rooms are set up as the best compromise for the kinds of sound that will take place in it.
Reverberation: How Much is Right for Music Venues?
Reverberation is the persistence of sound after the original source of it has stopped. It’s caused by multiple reflections from various surfaces, such as walls, furniture, etc. These multiple bounces of the same sound in essence reflect the same sound multiple times. While both reverberation and an echo describe a sound that persists after the source stops, an echo is a little different from reverberation in that it’s a distinct reflection of the exact source sound while reverberation is reflecting different bounces from various surfaces of the source sound.
Reverberation is multiple echoes emanating from reflective surfaces in the listening space. These reflections build up as the “echo” bounces from one hard surface to another gradually being absorbed over time by other absorbing objects in the space. It’s the reason the sound from an event held in a gymnasium or school auditorium that you hear is so much different than the original source voice or music.
The key to reducing reverberation is to add sound-absorbing materials to the space. The more absorption, the less reverberation will occur. The sound will decay much faster as we increase the sound absorption properties of the room.
How Much Reverberation Do You Need?
Reverberation is measured by a parameter called reverberation time. It’s usually defined as the length of time it takes for the original sound to decay to about 60 decibels from the initial level. A longer time delay impacts what the listener hears. Instead of hearing the direct sound waves, the listener hears both the direct sound and the reflected sound waves. Excessive reverberation causes the sounds to run together, causing the original sound to become muddy and garbled.
Reverberation is not bad. Music often benefits from differing amounts of reverberation. The right reverberation time can dramatically enhance the sound quality of performances. Music venues prefer longer music reverberation time because the reflected sound waves help blend the music in a more pleasant way as it reached listeners’ ears.
Acoustics are not only important for musical performances, but are important for venues that host weddings, corporate events, private parties, tradeshows, etc. In these cases, the source of the sound is the conversations taking place among the guests attending the event. Because of the volume of conversations and the fact that people are facing many different directions, the venue needs to treat more areas of the event space with sound-absorbing material. Shorter reverberation times are important in speech-oriented spaces. The less reflected sound waves makes it easier to understand the speaker clearly.
Unless your music venue was designed from the ground up with acoustics in mind or is a large open space, you should make every effort to optimize the acoustics of the space for concert sound. This is especially important when concert organizers turn another kind of space, e.g., bar, restaurant, parking lot, warehouse, etc. into a concert venue. It’s worth the investment of an acoustic engineer to help you optimize the acoustics for your performance space. You want a place where people will keep coming back because they love the sound at your events.
Concert Audio Setup: The Sound System
A concert sound system receives microphone and musical instrument inputs, processes and amplifies them and then outputs them through the speaker system. For concert audio, the main components of a sound system consist of speakers, amplifiers, and a sound board.
Speaker Issues for Concert Sound
- Frequency Response,
- Amplitude or Sound Pressure Level (SPL)
Frequency measures the number of sound vibrations in one second. It’s measured in Hertz (Hz). This relates to how low or high pitched the vibrations are. The more the vibrations (measured in Hertz), the higher the pitch sound is. We use both Hz and KHz (1000 Hertz) to show frequency ranges.
We hear by sensing the vibrations generated by sound waves in the air or other environments. The middle and inner ear translate these air vibrations into signals our brains can interpret. The normal human hearing frequency range is 20 Hz to 20,000 Hz or 20 kHz. This is the range for a healthy young person. The younger you are the better you hear. Infants can actually hear frequencies slightly higher than 20 kHz. As we age, we lose hearing first from the upper frequencies. A middle-aged man is more likely to have an upper range around 14 – 17 KHz range.
Low frequency noise, the frequency range from about 10 Hz to 200 Hz is especially problematic for many people. They are especially sensitive to such low frequencies. While a bass’ range may be from 31 Hz to 400 Hz, most bass playing occurs between 40 Hz to 200 Hz. The bass and the kick drum tend to play during most of a song and they share a lot of frequency ranges, especially 50 to 100 Hz.
A full range speaker has a frequency response of 60 Hz to 18 KHz and is fine for many musical instruments and genres of music. For genres where the kick drum and bass are key, you will need to complement them with subwoofers for the additional punch required. Subwoofer extends the frequency response down to 45 Hz or lower and will allow the full-range speakers additional headroom and increased output. For bands using a five or six string bass, the subwoofer speakers may need frequency responses down to 31 Hz.
Sound Pressure Levels (SPL) or Amplitude
Amplitude measures a sound’s pressure or forcefulness. The more amplitude a sound or noise has, the louder it is. We often refer to this as gain or volume. The unit of measure of sound pressure level is the decibel (dB). Decibels use a logarithmic scale. That means that as the number of decibels increases, the sound level increases rapidly or exponentially. We will confine the use of the decibel to sound pressure levels although it is used to measure the intensity of other things as well.
While the “normal” hearing audible amplitude range of a healthy young person is from 0 to 180 dB, anything over 85 dB has the potential to damage hearing depending on the frequency and length of time exposed to those sound levels.
The logarithmic nature of the decibel scale means that a very large range can be denoted by a useful number. For example, every increase of 10 dB on the decibel scale is equal to a 10-fold increase in sound pressure level (SPL). Near silence is expressed as 0 dB. A sound measured at 10 dB is 10 times louder. If a sound is 20 dB, that’s 100 times louder than near silence.
To recap, the difference is the sound coming out of a speaker at 90 dB is ten times louder than the sound coming out of a speaker at 80 dB! For example, 120 dB SPL may be clearer than “a trillion times more intense or louder than the threshold of hearing” at 0 dB.
A normal conversation is about 60 dB, a loud lawn mower may reach 90 dB and a loud rock concert can reach 120 dB or more. The effect of loud noise on hearing varies among people. The effect may vary by the frequency of sound to which they are exposed as well. In general, sounds above 85 dB are harmful, depending on how long and how often you are exposed to them. Such exposure may lead to hearing loss. Most occupational workplace noise regulations prevent employees from being exposed to noise greater than 85dB over the course of their workday. Leisure noise such as found at concerts is less studied and regulated.
The loudness of a sound is also determined by the sensitivity of the ear. The human ear is more sensitive to some frequencies than to others. The volume we receive thus depends on both the amplitude of a sound wave and whether its frequency lies in a region where the ear is more or less sensitive.
Children under 13 are even more susceptible to hearing issues with loud noise. That’s because they have better hearing than adults. It’s very important that hearing protection measures be taken for this group when exposed to high concert sound levels.
Concert Speaker Setup
The type of speaker setup that you will need depends significantly on the size and needs of your event. In a more intimate venue, you may only need two full-range speakers and a mixer setup. At a larger venue with tens of thousands of people in the crowd, you may need 40 speakers or more, with subwoofers as well. Most outdoor events are in a relatively open space, meaning few walls to cause reflections. This can be a benefit, providing the potential for better audio quality. The drawback is that you need a more powerful audio system to cover the same area.
The sound pressure level (SPL) of a speaker is a measure of how loud the speaker is at a given distance. The sound pressure level will be reduced by 6 dB with the doubling of the distance from it. So, a speaker with 135 dB output at 1 meter would have an output of 129 dB at 2 meters and a 123 dB output at 4 meters and so forth. 1 Meter is used, because that is the standard for Maximum SPL of a speaker in the specification sheet.
To maximize the SPL, you can add additional speakers. Each additional speaker will add 3 dB output. By adding another speaker to a 135 dB peak out speakers, you increased the output to 138 dB.
In a nutshell, the SPL output and number of speakers will determine how loud the sound will be at various distances from them. Remember that practically speaking, a one dB power gain is approximately 26%; a 3 dB is roughly twice the power gain, and 10 dB is about ten times the power gain.
Most loudspeakers have a sound projection pattern or dispersion pattern. Dispersion refers to waveguides, a portion of the speaker, usually a horn-like shape, primarily attached to the high-frequency driver or some boundary effects sometimes integrated into the speaker cabinet design. The waveguides do what you can probably guess by the name. They guide the sound waves to determine the coverage pattern of the speaker.
Low frequencies are omnidirectional, so when we set a loudspeaker on the floor, we effectively halve its radiating space at low frequencies. This produces an additional 3 dB of output (double the power) in the remaining coverage space. Speakers with a broad dispersion pattern create a larger sound field and a more consistent listening experience. A broader dispersion also means the speakers are more forgiving with where you place them.
How wide an areas a speaker will cover versus how high it covers is generally related to the physical size and shape of the speaker itself. However, sometimes speakers can have a variety of different coverage angles.
Dispersion is the direction the sound is projected both horizontally and vertically from the speaker. Knowing the speaker dispersion is important to determine speaker placement. The placement goal is to provide coverage to the entire audience. Today’s speakers allow for many speakers to be placed together in a tight-knit group or array, which allows for coupling with reduced interference between speakers.
The vertical dispersion of the speakers will determine how high the full-range speakers will need to be placed to provide the right sound coverage for the audience. There are several ways to configure the height needed for coverage including stacking speakers, utilizing speaker stands, scaffolding, or using trussing to achieve the height required to provide needed coverage to the full audience.
The higher the speaker, the further the sound will travel. It it’s too high, there will be a loss of impact in the front as the sound goes over their heads. If the speakers are not high enough, the sound may be uncomfortable for the people immediately in the front of the stage.
The main concern with horizontal coverage is ensuring you reach all the audience on both sides of the stage with minimal overlap. The more the speaker drivers overlap, the more likely you will produce phasing from sound wave interference. Phasing can cause hotspots, areas that are louder than other areas, or cancellation, where there is not enough sound.
Vertical Dispersion Horizonal Dispersion
An audio amplifier is a device that turns the low voltage signals from your source equipment received via line or mic inputs into a signal with enough gain to be used to power speakers. There are two functions of amplifiers used in a concert sound setup. The first includes collating and selecting multiple inputs and selecting the required level of gain and is generally referred to as the preamplifier. The second does the ‘heavy lifting’ and adds the gain to the signals to be used to power speakers. This is the power amplifier.
Good quality amplifiers don’t change the frequency of sound, i.e., the wavelength. That’s the job of an equalizer. What amplifiers do is adjust the amplitude of sound waves, i.e., the distance between the wave peak or trough of the wave and the position at rest. The size of the amplitude determines how loud the sound is. A larger amplitude means a louder sound, and a smaller amplitude means a softer sound.
When selecting amplifiers for a concert sound setup, the important specifications to consider include:
- Desired power output or SPL
The desired output to fill a venue with loud, clear sound depends on the type of music being performed as well as the venue space acoustics. Different applications require different ranges of power based on the desired loudness, speaker continuous power rating, and the loudspeaker’s sensitivity.
The higher the sensitivity rating of a speaker, the louder it will play with a certain amount of wattage. Loudspeakers with high sensitivity need less power than loudspeakers with low sensitivity. Large speakers used for concerts have a typical sensitivity of 110 dB.
- Signal to Noise Ratio:
Signal-to-Noise Ratio, or SNR, is essentially a measure of clarity. It tells us how much clearer the music is compared to any background noise in the system. It compares the level of signal power to the level of noise power. It’s most often expressed as a measurement of decibels (dB). Higher numbers generally mean a better specification since there’s more useful information (the signal) than unwanted data (the noise). The higher the SNR, the less interference you’ll have from background noises like hums, hisses, or pops. Concert amplifiers should be in the 100 dB SNR range.
- Total Harmonic Distortion (THD)
Total Harmonic Distortion compares the input and output of audio signals, with the difference in stages measured as a percentage. The percentage represents the harmonic distortion or deviation of the output signal. Lower percentages are better. Each component of the audio system adds some level of distortion, so it’s prudent to assess numbers to maintain audio output purity. However, Total Harmonic Distortion is not as important a specification these days since most values are often less than 0.005 percent.
- Nominal Impedance:
Think of impedance as a complex form of AC resistance. In the ideal situation, the impedance of the speakers and amplifiers should match to transfer the maximum power from the source to a destination speaker. This situation refers to as having matched, or balanced, impedances Matched impedance provides the optimum amplifier power level which is 50% for the total power output required. Mismatched impedance can lead to excessive power use, distortion, and noise problems.
How Much Power Do You Need?
Crown Audio has a calculator that determines the amplifier power required to achieve the desired SPL at a certain distance. It also accounts for the number of dB of amplifier headroom needed for audio peaks. The following link goes to Crown’s power calculator: Calculator
A high-quality amplifier used for concert production with matched impedances will provide the power needed (gain) while minimizing noise and distortion while delivering a linear frequency response, i.e., every frequency is equally represented at output needed.
The Role of the Sound Board
The mixer, equalizer, or sound board is the hub or traffic director of all the audio components of a performance. It’s a device that combines and routes multiple inputs to a common output. Inputs include microphones, line-level instruments, RCA components like DJ turntables, and other auxiliary sound sources. A professional mixing console ranges from 8-124 inputs, often called channels.
Why so many? In live concert sound you may have many instruments and vocalists and that’s why many inputs or channels becomes a necessity. A guitarist, a bass, a keyboardist, each vocalist takes a separate channel. A drummer usually takes up a channel for each drum and cymbal, averaging 8 inputs alone for a drummer. Try mixing a large band or orchestra in and you can quickly see where 124 inputs will come in handy!
Each individual input channel can be live or muted with extremely detailed controls of how that specific channel sounds and where it is output (to which speakers).
The sounds board’s role is to:
- Provide the necessary connections for various sound sources through its input channels.
- Allow the sound engineer to blend, or “mix,” all audio signals together and route them to a main output, i.e., speakers for a concert.
- Act basically as a level controller or equalizer for individual frequencies. It makes frequencies quieter or louder.
Sound Board Placement for Concerts
The front of house (FOH) equalizer or sound board has flexibility when it comes to placement. Generally, it’s placed between the speaker system so the FOH sound engineer can hear both speaker arrays equally. Some offset the console to the side of one of the speaker arrays to provide better viewing for the audience.
The FOH mixing console should be placed from two to four times the distance from the stage as the distance between the speaker arrays. For example, if your speaker arrays are 50 feet apart, then the console can be placed between 100 and 200 feet above the audience so that the audience doesn’t block any of the sound from the sound engineer’s ears.
The placement of the FOH console is also impacted by the type of music being mixed. For example, the console for a folk music performance will be placed closer to the stage than one for a heavy rock outdoor performance. Be aware that these are only guidelines, and you will find FOH mixing consoles at various locations at different events.
Equalization (EQ): Obtaining the Best Sound Quality During Concert Audio Setup
During sound check before a performance, the various audio sources from instruments, microphones, and auxiliary devices are assigned an input channel on the sound board. Each of these input channels are then manipulated to provide quality sound to the audience.
Equalization is the process of adjusting the amplitude or gain of specific frequencies to allow the audience to hear the sound the way the artist intends for them to hear. An equalizer or sound mixing board is basically a level controller for individual frequencies. It makes frequencies quieter or louder.
Most musical instruments and even human voices are defined by the range of notes they can make, thus, for instance, a female soprano is expected to be able to output maximum power (frequently also referred to as singing) in a certain pitch range This range corresponds to 262 Hz to 1047 Hz although some sopranos can go lower and higher than this range. So, if we want to pull out a soprano voice from the background these are frequencies we would concentrate on when equalizing. see here if you want to learn about frequency ranges of musical notes and instruments.
Fundamental Concert Input Frequencies
Equalization also is used to address problems such as problematic frequencies that can be annoying or clearing up the muddiness that occurs when instruments with overlapping frequencies clash with each other, making it difficult to perceive simultaneous sounds.
This is only a basic introduction to the much more complex subject of equalization
The audio setup for concerts is a complex undertaking, matching the audio system’s components to the size and acoustical properties of the performance venue, the size of the audience, the distances involved for the sound to reach its furthest members, and the nature of music being performed. The resulting sound is the result of having the right speakers, amplifiers, and sound board equipment for the event and the skill of the sound technician in equalizing the various frequencies to match what the artists’ want the audience to hear, and the acoustical properties of the venue. When these come together, the audience will hear the best sound possible and will enjoy the music as it was meant to be heard.


The ability to communicate across multiple devices at any time, irrespective of location is a ‘must have’ for many organisations in an effort to reduce costs and increase productivity. Making a voice call from a desk phone, sending an instant message to a remote colleague, making a video call, sharing presentations to collaborate and checking presence. Add the ability to do all of this on your mobile device too and you have a unified communications solution.
Unified communications provides all this functionality from one manufacturer. This ensures users don’t have to learn multiple different platforms from different vendors, instead needing to use only one easy to use application. So, whether you’re working in the office, Kyrgyzstan or Timbuktu, a unified communications solution from ScanSource Imago will improve communications within any organisation; improve efficiency and productivity while reducing operational costs.


The speaker reproduces the sound effects of the gun. We recommend a 50
mm or larger speaker, to reproduce the sounds generated by the PCB (6)
with decent volume and clarity.
4 or 8 ohm speakers can be used on the FragTag mainboard. Preferably
the speaker should have a cone made of Mylar, as this provides better
water resistance than paper cones. We still have had good success with
paper cones so this is not a hard requirement.
The speaker should be mounted behind a grill, to let the sound out while
still protecting the fragile cone from damage.
Some example speakers:
63 mm mylar
53mm paper cone (keep it dry!)


لایسنس نرم افزار Nemo Windcatcher V188.8.131.52
قابلیت لاک بر روی هر سیستم
هر لایسنس برای یک سیستم میباشد و بر روی آن سیستم لاک خواهد شد
پس از خرید بخش فروش تمز ایران با شما تماس خواهند گرفت و لایسنس مربوطه و همچنین آموزش های لازمه را در اختیار مشتری قرار خواهند داد
Nemo Windcatcher V184.108.40.206 Version https://bit.ly/2kw98Ri Download Link
Key Features & Specifications
Optimize OPEX by streamlining analytics processes in a multivendor environment
Vendor-neutral, multi-data analytics solution that provides the most comprehensive network analysis available in the market.
Supports a wide range of data inputs, e.g. drive test data, including 5G NR, and Operations Support Systems (OSS) data
One tool to process data from various tools from major measurement vendors
Industry-leading desktop-based drive test data post-processing tool for radio access network analysis
For network planning engineers, network performance engineers, handset testing groups, and technology development groups
Available as a client server and stand-alone desktop configuration for drive test use case automation and drill down analysis
Nemo WindCatcher Server automates key drive test use cases, such as acceptance, benchmarking, optimization, and monitoring
Full automation for data management, processing, and reporting
Data can be downloaded from Nemo Cloud, FTP sites or uploaded from clients
Near real-time email reports
Interactive dashboards with ability to drill down
Customizable reports and dashboards
Multiple clients can be used for standalone analytics or connected to the server to access processed data, reports, dashboards, or raw data
Easy to use, versatile
Single-click option to load large amounts of data files
Drag-and-drop based UI with dashboards for easy overview
Processes large volumes of drive test data and/or walk test data with ease
Multi-core processing functionality for faster data output and lesser wait times
Multiple devices and data sets can be merged to create large data sets
Designed for optimization, acceptance, and benchmarking of new technologies and high speed data networks
Support for latest wireless technologies, such as 5G NR
Wide application support
Network analytics in a multivendor environment
Nemo WindCatcher is the industry leading desktop-based drive test data post-processing tool for radio access network analysis.
It is a vendor-neutral, multi-data, multi-wireless access technology solution that provides the most comprehensive solution available in the market. It supports a wide range of data inputs and all major data collection tools on the market, and it is optimized for heterogeneous test environments and processes.
Nemo WindCatcher is loaded with extremely robust out-of-the-box features for powerful data aggregation and analytical functionality, one-step analysis, and detailed reporting. Users can quickly identify and solve problems maximizing overall productivity and efficiency.
Flexible in-depth analysis
Wireless network operators can leverage Nemo WindCatcher to analyze large amounts of drive test data to enhance customer experience through efficient network optimization, new technology roll-out, and troubleshooting. Operators can deploy new technologies, such as 5G NR, LTE, LTE-A and Voice over LTE (VoLTE), using the advanced analytics in Nemo WindCatcher. Device manufacturers can use Nemo WindCatcher to modify the functionality of their devices for use in different network conditions.


Engineers, OEMs and integrators have struggled for years to balance the cost of PLCs with their I/O needs, and Turck's new ARGEE technology solves this challenge. ARGEE empowers manufacturers to add logic to compatible I/O devices without a PLC, driving down the cost per I/O point and increasing flexibility in control.
The ARGEE environment is accessible via an HTML5-compatible web browser, which eliminates the need for complex third-party software or licensing that is common with PLCs. Inside the ARGEE environment, users can set condition and action statements to be carried out by the I/O device. The application's intuitive design includes a flow chart - similar to ladder logic - which allows for easy programming by users of all experience levels.
By taking advantage of ARGEE on Turck's multiprotocol Ethernet I/O devices, manufacturers are able to realize a better control solution - field logic controllers (FLC).
FLCs benefit small to mid-size OEMs and manufacturers who are looking to add logic without investing in a PLC. FLCs can be used in one of three ways:
- The FLC stands alone: The FLC performs all logic as a standalone application without a PLC
- The FLC backs up the PLC: If the FLC loses communication with the PLC, the FLC can take over and either shut down or maintain the process.
- The FLC and the PLC work together: The FLC can monitor an application and send defined variables and updates back to the PLC.
"With FLC powered by ARGEE, engineers can add control capabilities and I/O points to applications without having to invest in or upgrade a PLC," says Randy Durick, Vice President of Fieldbus Technology at Turck. "Because the I/O devices carry advanced ingress ratings, logic can move outside the panel and onto machines for local control and real-time performance monitoring."
Specific features of ARGEE include:
- Toggling bits, the utilization of timers and counters, performing arithmetic functions, monitoring of program status via the built-in HMI capabilities and communicating with an upper level PLC
- HTML5-based programing, which makes the environment accessible on any device via capable web browsers such as Chrome and Firefox
- Integration with digital, analog, IO-Link, RFID, and specialty I/O types
ARGEE technology is standard in all Turck multiprotocol Ethernet I/O devices. These devices can be used on any PROFINET, Modbus TCP, or EtherNet/IP platform, and include the TBEN-L/TBEN-S on-machine block I/O, BL compact on-machine flexible block I/O, and FEN20 in-cabinet block I/O.


Http_ whatismyipaddress.com ip-lookup
go to an IP address tracking website, such as whatismyipaddress.com, server will check with the actual IP that's being spoofed to make sure its r 5 Jun 2012 feature in Google AdWords. Check it out. You can determine your own IP address at WhatIsMyIPAddress. To determine a competitor's IP 5 Nov 2015 Hi all, http://whatismyipaddress.com correctly identifies the country (UK) and county of my IP Any suggestions please or more generally is piwik IP lo… This the default thing in IP lookup which all IP lookup sites s Enterprise IP Location by Kloudend, USA is built on AWS & trusted by Fortune 500. Supports JSON, CSV, XML in PHP, JS, Java, Python, Go, Ruby & C#. Bienvenido a whatismyIPaddress.com. Su dirección IP es algo en lo que probablemente piense muy poco, pero es de vital importancia para su estilo de vida en Getting a random threatening email from this IP address out of nowhere, really relieved I hope you like the WhatIsMyIPAddress.com redesign and how we've Location: Madrid, Spain - 22.214.171.124 is a likley static assigned Corporate IP address allocated to M247 Ltd. Learn more.
Escuela de Ciencias de la Computación - UTPL
IP Address location information shows country name, region, city, ISP, TimeZone, Currency, Dailing code etc. The I nternet P rotocol Address (or IP Address) is a unique address that computing devices such as personal computers, tablets, and smartphones use to identify itself and communicate with other devices in the IP network.
My-addr.com - 49 páginas similares a My-addr
Hello Asia. The IP lookup tool can only give you an estimate of the current location of the IP address you’re looking for. IP stands for Internet Protocol. This protocol, like many others such as HTTP, TCP, UDP, etc., is responsible for establishing communications in most of our networks. An Internet Protocol address (IP address) is a numerical label assigned to each device connected to a computer network that uses the Internet Protocol for communication.
Seguridad en servidores MTA sobre Linux - Universidad .
Without you even noticing, your IP address is used when you open a … WhatIsMyIPAddress.com, Tustin, California. 225,273 likes · 29 talking about this. IP address lookup, location, proxy detection, email tracing, IP hiding What is my IP? Get your current public IP address 03/11/2011 What is IP8.com. IP8.com helps you stay safe online by learning about which of your personal information is accessible.
IP Address Lookup for 126.96.36.199 in San Jose United States
ip address traced to : 188.8.131.52. I live in Elk Grove Village, IL and my provider is Comcast. He now thinks I'm doing sneaky stuff on the side. I am NOT and I didn't do anything.
Crear radio online con Shoutcast en Ubuntu, Fedora y otros
25/2/2021 · WhatIsMyIP.com® is the industry leader in providing IP address information. Moreover, we provide tools that allow users to perform an Internet Speed Test, Location Lookup, Proxy Detection, Whois Lookup, and more. Extensive tutorials that show users how to trace an email address are also available. ABOUT WHATISMYIP LOOKUP Ok, so this page shows you your public IP Address. But the MxToolBox WhatIsMyIP also runs a blacklist check on over 100 lists to tell you if you are blacklisted.
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Soon afterward, he shared his website and IP lookup solution on the web. 04/03/2021 Easiest is to type the question “What’s my IP/whatsmyip.com” in the search bar of your favorite web browser. We will clearly show you your IP on our front page. We will also try to locate you based on … Whatismyipaddress.com Single IP Lookup. fugazy.
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Cisco has said that Nigeria needs between 30,000 and 70,000 Information Technology workers to bridge the gap in networking professionals.
The networking company, however, lamented that tertiary institutions in the country were not producing enough Information and Communications Technology graduates to meet this demand.
It stated this after a meeting of Cisco Networking Academy instructors at the University of Lagos.
The General Manager, Cisco, English – West Africa, Mr. Dare Ogunlade, told our correspondent that governments also urgently needed to deploy policy and training programmes if the gap in networking professionals must be bridged.
He said that should governments fail to do this, “the global shortage of skilled Internet providers (networking professionals) would be at least 1.2 million people in 2015.”
The INSEAD Business School’s ‘Global Talent Competitiveness Index 2014’, co-authored by Cisco, supported the statistics.
According to the latest Cisco Visual Networking Index, ‘Global Mobile Data Traffic Forecast: 2014-2019 forecast’, mobile data traffic is expected to grow 11 folds over the next five years in Nigeria, which is a compound annual growth rate of 63 per cent – two times faster than the expected fixed IP traffic growth.
This highlights that connectivity is accelerating at a fast pace in the country, thus creating a growing need for skilled ICT professionals.
Cisco said that the lack of young ICT professionals was not limited to Nigeria. “Increased connectivity, the Internet of Everything, rising digitisation of all business activities, globalisation of trade and travel, and economic growth globally has created the same problem in a number of countries globally,” Ogunlade said.


NSTAR Restoration Efforts
Our advanced, computerized control system often gives us first notification of a large power outage. We also get reports from public safety agencies and local police and fire departments.
NSTAR also depends on customers to report outages - either by using our toll-free number (800-592-2000) or our "Report an Outage" online tool. Smaller outages can sometimes go undetected by the automated parts of our system.
Our specialized computer system allows dispatchers to analyze and take care of many problems electronically, which can reduce the duration of an outage. If the problem cannot be corrected in the control room, a crew will be dispatched.
In the event of a major storm, NSTAR implements an emergency plan of operation which places the efforts of the entire company behind power restoration 24-hours a day. We will try to restore electric service to the maximum number of customers in the shortest possible time.
NSTAR is in constant communication with government representatives, town officials and the media throughout any outage; therefore you may be able to receive updated information through your battery-operated radio.


- AMS is a state-of-the-art switchgear developed by German experts.
- AMS is designed for full ratings of medium voltage switchgear, rated voltage ranged from 7.2kV up to 40.5kV.
- AMS is a fully metal-clad, air insulated, arc-proof and compact design switchgear.
- AMS is equipped with next generation cassette type of withdrawable embedded pole vacuum circuit breaker.
- High speed earthing switch with short circuit making capability is employed in AMS switchgear.
- The mechanical strength and anti-corrosion properties of the structural frame of AMS are ensured by ALUZINC sheet metal with double bending and riveting fixing technology.
- Optimized electrical field and electrode screen technology are adopted to achieve high insulation reliability of AMS.
- AMS has a cable compartment with ample space and the cable connection height is up to 800mm.
- AMS is designed with a perfect interlocking system which provides highest safety for operating personnel and equipment itself.
- AMS has inspection windows for both breaker and cable compartments, through which, the position of withdrawable unit of the breaker and the situation of the cable connection can be observed easily.
- AMS can be equipped with traditional or integrated protection devices, as well as traditional CTS and VTs or sensors.
- AMS is safe and reliable switchgear which had been widely accepted by various customers around the world.
|2||Rated Frequency Withstand Voltage||50KV|
|3||Rated Impulse Withstand Voltage ( BIL)||125KV|
|4||Rated Busbar Current||UP TO 2500A|
|5||Rated current for Feeder||UP TO 2500A|
|6||Rated short time Current/ time||25KA/3s|
|8||Degree of protection||IP4X|
|11||Width – mm||(800)1000|
|12||Depth – mm||1680|
|13||Height – mm||2250|


Whats the difference 60w 80w at 2mJ compare to a 100w at 1.5mJ jpt fiber laser source
Whats the difference 60w 80w at 2mJ compare to a 100w at 1.5mJ jpt fiber laser source?
In the Case of a MOPA with Variable Pulse Width the Max Energy happens at the lowest Frequency. For the 80 it's 60KHZ-40KHZ depending on the Pulse Width. The 100 it's 75KHZ-67KHZ, the Wattage Rating is over the entire Frequency Range, So the 80 at 2mJ per Pulse will have less Energy in the higher Frequencies than the same 80 Watt Module that output 1.5mJ. It is highly likely that the 80 Watt will output less 1.5mJ at 75KHZ like the 100 Watt, the ability for the Module to deliver high Pulse Energy decrease with Frequency (1000 Cycle per Second).


Windows Speech Recognition
For Windows users new to text-to-voice, Windows very own speech recognition is a great starting point.
To Enable Windows Voice Recognition tool click Start > Control Panel > Speech Recognition Options
DSpeech is a TTS (Text To Speech) program with the functionality of ASR (Automatic Speech Recognition) integrated. It is able to read aloud the written text and choose the sentences to be pronounced based upon the vocal answers of the user. Allows you to save the output as a .WAV, MP3, or OGG file.
Jott turns voice into text, send emails and text messages, set reminders, organize lists, and post to web services and business applications – all with their voice, using any phone. Jott also converts voicemail into email and text messages, making voicemail a more productive


Since updating windows internet wont connect
If you have another phone or computer or any device that can see Wi-Fi connections make sure that the other devices can see and connect to it.If you have lots of connections to the Wi-Fi already then disconnect a few of them.The second thing that you can try, to fix your P8 Wi-Fi connectivity issues, is to turn your P8 and/or your router off and then turn them on again.Read a bit more on what you need to do below as this sometimes resolves the issue.I am sure that you will be able to resolve the connection issues with one of them.
Authentication issues are often (not always) a problem with the password.
You shouldn’t need to press the reset button, just pull out the power cord from the router or modem, count to 15 secs, and then plug the power cable back in, wait a few minutes for the device to boot back up then try it again. Unplug the power cable count to 15 and plug it back in.
If it’s still connecting but not using the internet then power cycle the phone itself, turn it off and then turn it back on.
Whenever I want to share an Internet connection through Wi-Fi and turn my Windows machine in Wi-FI access point, I find myself googling again because I somehow can’t memorize the procedure.
Google is a great brain extension for people with a bad memory like me.
Search for since updating windows internet wont connect:
However, in this case, it always leads to frustration because many guides about Internet Connection Sharing (ICS) are incomplete, describe the procedure in the wrong order, or confuse different Windows versions.


The infrastructure application provides a number of features to make it easy to run pipelines. These include automatic infrastructure provisioning, secure data transfer, secure connections to the cloud, and a simple interface for monitoring all the infrastructure in use (including non-OmicsPipelines infrastructure).
📄 Getting Started
Get set up with the OmicsPipelines desktop application
📄 View Infrastructure
View the infrastructure in your account.
📄 Provisioning Cloud Infrastructure
Create a virtual machine using the desktop application.
📄 Connect to the Dashboard
Connect to your virtual machine using the desktop application.
📄 Manage Infrastructure
Manage your infrastructure using the OmicsPipelines desktop application


Smart buoys and sensor technology paired with software solutions will serve the next generation with a grid of possibilities. Our everyday lives are already driven by IoT (Internet of Things). It is now time to build the Internet of the Ocean.
The durable buoy paired with advanced IoT technology allows for around the clock surveillance of your gear. Never lose track of your assets at sea.
The Sunfish buoys will sustain the pressure even if submersed to 200m below sea level. Once surfaced, it will continue transmitting its position.
The smart buoys gives the fisherman an opportunity to survey the position of his fishing gear. In combination with our software, Digicatch, the buoys are “smart” and have lots of useful features. One of these features will send a push warning to the fisherman’s mobile phone or his onboard navigation system if there is something to be aware of. For example, if it is overdue for the gear to be harvested or simply that it is moving out of position. This way, the fishermen can take actions to prevent the loss of his gear.
We are developing advanced sensor systems and software solutions for monitoring fishing operations, aquaculture, metocean conditions and subsea operations.
Our technology is based on wireless acoustic transmission to the surface that enables the monitoring of operations and the environment in real time and the collection relevant data.
We are also developing software solutions to provide user-friendly front-end visualization of the collected data and to generate reports and analyses tailor-made for each user.
Our stationary trap concept is a new and high tech fishing technology that will work passive and autonomous. The traps may be placed near fish farms, offshore wind power installations and other vulnerable areas. This technology is an important contribution to make fisheries and fish farms more sustainable.


Easemind Technology Ltd. is established by a team of engineers and programmers specialized in industrial and telecommunication fields. We have been in business of hardware and software development of wireless data logging devices and industrial telemetry equipment since 1989. The business and marketing concept of the company consists of: * Development according to the actual market needs and customer demands, * Utilization of reliable, trust-worthy and right-to-the-edge technology, * Turn-around product delivery and service fulfillment within time limits, * Cost-effective solutions: the right price for the right product, and* One Stop Solution of hardware, software and user applicationsEASEMIND is now a group of companies specialized in RF wireless sensor & mobile data tracking devices for consumer and industrial sectors. Our sales offices are located in US, Hong Kong and China with manufacturing facilities in Zhejiang and Shenzhen, China. OEM or Distributors are welcome! Contact us if you are interested in being our regional distributors.


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- Universal fit
- “Hey Voice Assistant” always on
- Up to 4.5h of listening time (on one charge)
- More than 24h of listening time (with Wireless Charging Case)
- Around 1h of listening time on only 5 minutes of charging
- Wireless Charging Case
- Active Noise Reduction
- Hear Thru mode
- Dual optical sensors
- Force sensor
- Sweat and water-resistant (IPX4)
- Guaranteed Safe Checkout
- Choosing a selection results in a full page refresh.
- Press the space key then arrow keys to make a selection.


In recent years, there has been a boom in ed-tech services, with numerous new companies promising career-building services aside from the already existing ones such as Coursera, Khan Academy, Byju’s, etc. However, one of the toughest hurdles to cross has been satisfying one of the basic requirements for accessing these services. Approximately 4 billion people worldwide struggle to adequately access the internet, especially people in rural areas of developing and underdeveloped nations. As such, ensuring access to the internet via affordable devices is the need of the hour.
Over the last few years, the satellite internet industry has gained momentum. While previously companies such as Teledesic had struggled to achieve sufficient returns on their investments, recently companies such as SpaceX, OneWeb, Google and Airtel have been vying to capture the market in various regions. While SpaceX’s Starlink uses a network of satellites to provide internet services over large areas, Google Loons uses high altitude balloons to do the same. Satellite internet solves the issue of decreasing network speeds due to increasing distance from wireless cell towers. Companies tend to build less towers in areas with low population density and low buying power, which effectively ensures that people in remote areas would be plagued by the lack of network connectivity and low internet speeds. Satellite internet due to its higher altitude can provide internet services over a larger radius. This would help students in rural areas to access both live and pre-recorded online lessons seamlessly, thus reducing the gap in educational quality between students in rural and urban areas.
The other major drawback of online education has been the lack of affordability when it comes to the educational devices such as laptops and smartphones. High income inequality in underdeveloped areas of the world means that a large section of the people still lack buying power when it comes to laptops and tablets. However, it might be achievable to secure smartphone access for most of the people due to the falling prices of smartphones worldwide. While previously Lithium-ion batteries used to be the norm for power storage in smartphones, recent advancements in Sodium-ion batteries could offer a cheaper alternative, bringing down the price of smartphones even further. In early 2022, a subsidiary of Reliance Industries Ltd acquired Faradion, a maker of Sodium batteries. Various other Sodium battery producing companies have been started over the past decade. This signifies major businesses are viewing Na-ion as a potential alternative to Li-ion. This would further help to bring down the price of various battery powered devices due to the greater availability of Sodium.
These two relatively recent developments in the technological sphere could help alleviate concerns over the EdTech industry leading to greater inequality in the future. With more widespread access to educational devices and high-speed internet, children in rural areas could gain unhindered access to quality education, one of the basic requirements for achieving equality.


A few interesting points from a recent article on CRM daily: IP Systems have come of age. The need for truly integrated call centers, lower cost infrastructure and the lower costs of IP based systems are leading to a rise in the implementation of IP based Contact Center systems. Fully integrated systems that can handle multiple channels (Voice, Chat, E-mail and Fax) in a single queue and over different locations, instead of the loosely integrated, old, copperwire based PBX and TDM systems.
some interesting quotes from this article.
“That cut-over to having a majority of contact center seats IP-enabled could occur in the next few years. This trend, say analysts and suppliers, is being driven by legacy switch replacement cycles, adoption of IP by small/midsized contact centers, new sites, remote agents and informal contact centers, and by customer migration to text and e-mail from voice.”
These IP based systems are not always implemented as a replacement for TDM based systems, but sometimes serve as an addition, enabeling quick deployment of new call and contact centers.
“Raun Kilgo, Director of Product Management, Aspect reports that while there has been increased IP adoption among his firm’s customers it has not been, in most instances, as replacements for TDM. “
“Most of our customers have already invested in and are comfortable with TDM applications for their mission-critical inbound and outbound work and would need
a compelling event to rip and replace their existing infrastructure ,” reports
Kilgo. “Alternatively, IP is the ideal solution for companies implementing new
contact centers or deploying remote or at-home agents.”
Remote agents or the possibility to quickly scale up a call center in the event of an adverse event are benefits an IP based solution offers.
“Another advantage IP provides is reduced facilities and IT expenses through less wiring and no need for separate phone rooms: calls are increasingly handled via integrated, centralized and often off-site data centers. That also means faster, easier, and less costly expansion. New contact centers can be up and running in weeks, with no more sometimes lengthy waits for the telcos to install the lines.”
The real advantage of having an IP based solution, in my view, is something which could be dubbed the pervasive contact center. All your employees can virtually become part of the call center, so that experts are able to answer your customers inquiries, when an agent is not able to. Providing a correct and direct answer to a question, without the hated call back request is something should enable you to quickly increase customer satisfaction.
Call escalations, especially support calls to outside experts at regional or head offices or who are mobile or at home are easier to set up and have greater functionality with IP, provided it is coupled with unified communications.
In other words, IP based contact center systems hold a lot of promise for the present and the future, when combined with On Premise or On Demand based CRM applications can enable you to better serve your customers and achieve higher levels of customer satisfaction.


New York City is one of the world’s most advanced, connected cities in the world. A walk through Manhattan will reveal commuters, business professionals, students and artists with the latest in smartphones to help them stay connected with the fast pace of city life.
With smartphones so prevalent, it may seem surprising to even think about payphones or phone booths – a technology that is quickly going the way of the typewriter and telephone operators. However, during Hurricane Sandy, many New Yorkers without electricity to charge their mobile devices relied on payphones to make calls and check on family and friends. Phonebooths provide a vital source of communication during times of disaster, particularly if cell-phone towers are out of service.
NYC Phonebooths Getting An Upgrade
Phonebooths in New York City are getting an upgrade from coin-operated landline phones to Internet-enabled hotspots with community information, directions to landmarks and local retailers, and emergency assistance. The large, touch-screen displays are similar to a smartphone screen, and are used not only to make outgoing calls and receive information in times of emergency but provide helpful information on local retailers, the subway system and maps of the city.
Some questions remain about the physical durability and safety of the new, hi-tech booth. Tom Touchet, CEO of City 24×7, the company leading the project – which works with Cisco to provide the network technology which powers the LG Flat Panel displays – attested to their durability, noting that they have been installed in other high-traffic areas such as Newark, New Jersey without sustaining damage. The booths were operational in areas with power during the hurricane, providing a vital form of communication – displaying emergency information from City Hall.
Smartphone Boots to be Installed Across All 5 NYC Boroughs
City 24×7’s current plan is to install 250 “Smart Phone Booths” across all five boroughs of New York City, including high-density areas such as Times Square, Grand Central Station and World Trade Center. They also plan to expand to other major cities including Miami, Los Angeles and Boston.
While New York City offices might still rely on landline connections, it has become clear that the future will be dominated by smartphones. The innovation seen in these new smartphone booths will not only help travelers and residents navigate the city and find a bite to eat, but have already proven useful in emergency situations, including Hurricane Sandy and during a police incident in Times Square.


|Home||» Products||» Ladder Type Cable Trays|
Banking on the rich experience of our professionals, we hold strong expertise in manufacturing, exporting and supplying an wide range of Ladder Type Cable Tray. Extensively demanded in various lengths, designs and widths, this cable tray is used for supporting electrical cables and offers a non-conductive surface for cable to reduce shock hazard. It is designed and manufactured using premium quality basic materials and avant-garde techniques in conformance with international quality standards. To meet the specific needs of clients, we are offering provided Ladder Type Cable Tray in various lengths and widths at most competitive prices.
Resistant to corrosion


Shipsonic™ has different transducer classifications; water-resistant IP63 and water-proof IP67.
Our different transducers, often used in combination, are:
- 50 Watt 17 kHz, different housings available
- 50 Watt 40 kHz, IP67
- 80 Watt 23 kHz, different housings available
- 100 Watt 28 kHz, IP63
- 100 Watt 28 kHz, IP67
- 120 Watt 20 kHz, IP67
Correct ultrasonic frequency
Scope-reading of frequency response test
We use heavy, low frequency transducers for 2 reasons:
– the lower ultrasonic frequencies are more effective
– the resonance frequency of a transducer goes up about 16 kHz
once it is installed on an object like a box-cooler (thus a 20 kHz
transducer will make your box cooler resonate with approximately
Our design features a standalone software module for each transducer, looping constantly through all frequencies from 20 to 60 kHZ. This enables us to ensure that each transducer/object combination always receives pulses in its optimum range.


LeapSwitch Networks offers fully managed and self managed dedicated servers across the globe with the latest Intel/AMD processors and premium Tier 1 bandwidth. All our servers come with RAID capability for data security and IPMI for out of band connectivity. We have servers upto 256 cores , 2TB RAM and 24 HDD, SSD and U.3 NVMe drives capacity. Please contact us if you do not find the configuration that you are looking for. Our 24x7 support team is always here to assist you.
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A dedicated server is a type of web hosting where the user leases an entire physical server from a hosting provider, which is exclusively used by the user's website or application. This means that the user has complete control over the server's resources, including the CPU, RAM, storage, and bandwidth.
They offer several advantages over other types of hosting, such as shared hosting or virtual private servers (VPS), including: Performance, Security, Customization and Reliability.
Single CPU with default hardware configuration we aim to activate within 12 hours. For Dual CPU configurations, expected setup time varies from 1-3 days. Please contact sales members of our team and will endeavor to provide an ETA.
Our company manages all hardware, network, virtualization, software, operating system, and control panel issues. The client is only responsible for 3rd party applications and coding related issues.
Our company is responsible for hardware and network support and the virtualization environment, but the client will be responsible for installing, configuring and upgrading the operating system and other required software.
Operating systems available for dedicated servers are as follows:
Linux: CentOS 8.x, Ubuntu 22.x, Debian 11.x, Proxmox 7.x, AlmaLinux 9.x, CloudLinux 8.x and others.
Windows: Windows Server 2016, 2019, 2022 and others.
Others: Custom OS.
It is a standard inclusion with all our Dedicated servers.
Indeed, we provide a range of add-ons and additional services for our dedicated servers, like managed services, Acronis backup options, and custom configurations.
We provide 99.9% network uptime guarantee SLA.


Another useful utility app, WiFi analyzer is a great little app that helps you decide which channel is best for your WiFi.
From the CNET review “A Wi-Fi channel is a range of radio frequencies that a Wi-Fi network uses to communicate with wireless devices. There are a finite number of accessible channels, so when several Wi-Fi networks occupy a small area, channels will often get overloaded, and that’s bad.” Read more
This app comes in handy when you live in apartment, or when you discover that all of the routers in your workplace are set to the same channel.
Every tech nerd (or person who might be responsive for wireless networks) should have this app.


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This article discusses a network setup which might prove useful for people who like to put some extra effort into connecting their machines to the Internet. The goal is to build a secure client and server farm on a single IP address. Next to the clients and servers, the network contains two other components. The first one is a Network Address Translator (NAT), which makes it possible to connect more than one machine to the Internet using just one IP address. The second one is a Packet Filtering Bridge (PFB) that transparently filters IP traffic from and to the Internet.
The PFB machine - running OpenBSD 2.9 - is the only machine connected directly to the Internet. Two of its network interfaces are for the bridge itself (p1 and p2) and one is the management interface (p3). Interface p3 makes it possible for the PFB to output its logs to another server, or for an administrator to log into it to check its logs. This might not seem very secure at first sight, but the interface is actually protected as well as the clients on the LAN.
The NAT machine - running FreeBSD 4.3 - is connected with one interface to the PFB machine (n1). The other two interfaces are connected to the hubs of the clients (n2) and servers (n3). This seperation between a client and a server LAN has at least one obvious advantage. If a weakness is found in one of the services of a server, the clients are still relatively safe from attack. Otherwise, the clients would be on the same network as the compromised server and would have made an easy target for the attacker.
Let's start with some theory about Network Address Translators (NAT) and Packet Filtering Bridges (PFB).
Commonly, your Cable or xDSL ISP assigns you a very limited number of IP addresses (usually just one). You run into trouble if you want to connect more machines to the Internet than the number of IP addresses given to you. NAT is a technology which you can use to solve this problem to some extent. With NAT, multiple internal IP address (on the LAN) are aliased to a single public IP address (on the Internet). Requests from machines on the LAN to machines on the Internet are transformed by the NAT machine. It modifies the source address in the IP packet and sends the request in its name to the machine on the Internet. Once it gets a reply, it modifies the destination address in the IP packet to the client on the LAN.
Typically, a NAT machine has two network interfaces, one on
the Internet and one on the LAN. The public IP address on the Internet is
assigned to you by your ISP, so you assign this IP address to the interface
connected to the public network. But what about the private IP addresses on the
LAN? There are some IP addresses which are available for exactly this situation.
You can choose between three IP ranges:
Let's say you are using the third IP range. You would then assign one of the IP addresses in this range (e.g. 192.168.0.1) to the interface connected to the private network. Every other machine on this network is assigned one of the remaining addresses in the range. Here's an example of the complete IP configuration of the machines involved:
|IP address||Subnet mask||Gateway|
|NAT private||192.168.0.1||255.255.255.0||No gateway|
|LAN machine 1||192.168.0.2||255.255.255.0||192.168.0.1|
|LAN machine 2||192.168.0.3||255.255.255.0||192.168.0.1|
A bridge is used to connect two network segments to create a single, larger network. This is normally done to create the appearance of a single network when there are in fact two physically seperate networks. The bridge in this article is not just a simple bridge, but it functions as a packet filter as well. Devices that filter packets have rulesets that tell them which packets should be allowed and which should be blocked. Typically, a firewall is used for this job, but a disadvantage of this setup is, that the firewall is visible on the Internet. This makes it a great target to anyone trying to get into your network. If the firewall gets compromised, all machines within the LAN are in the open, waiting to be compromised as well.
In our case, the packet filtering is done by the bridge. A normal bridge takes all packets from the network on one interface and copies them to the interface on the other network and vice versa. Our PFB examines each of these packets before they are copied to the other interface. It checks its ruleset to see if a packet should be allowed to pass or should be blocked. Unlike the NAT machine, the PFB machine doesn't change anything in the packets themselves.
The IP configuration of the PFB and the machines on both networks is really simple. The PFB is not assigned an IP address at all. This ensures that machines on either side of the PFB are not aware of its existence. Because of this, the machines on both networks should be configured as if the two networks were really one larger one. The cool thing about this, is that the PFB can't be reached from the Internet, making it impossible to attack it on the IP level.
Now that we've seen the theory, let's shift our attention to the implementation. As mentioned before, we are using FreeBSD 4.3 for the NAT machine and OpenBSD 2.9 for the PFB machine. We won't go into much depth with regard to the client and server machines, although we will discuss the IP settings of these machines.
We want to be able to divert and filter IP packets. The FreeBSD generic kernel doesn't allow this, so we have to make our own kernel. We do this by making a copy of the generic kernel:
cd /usr/src/sys/i386/conf cp GENERIC NATThen we add the following lines to the NAT file:
options IPDIVERT options IPFIREWALLNow we can compile our new kernel and install it:
config NAT cd ../../compile/NAT make depend make make install
We should now edit some configuration files to really enable the capabilities just added to the kernel. The first file is /etc/rc.conf which contains the network interface configuration as well as the services configuration:
hostname="nat.example.com" # Host- and domainname ifconfig_xl0="184.108.40.206" # Network interface connected to the PFB (n1) ifconfig_rl0="inet 192.168.0.1 netmask 255.255.255.0" # Network interface connected to the client LAN (n2) ifconfig_rl1="inet 10.0.0.1 netmask 255.255.255.0" # Network interface connected to the server LAN (n3) defaultrouter="220.127.116.11" # Default router on network interface n1 firewall_enable="YES" # Enable firewall capabilities gateway_enable="YES" # Enable gateway capabilities natd_enable="YES" # Enable network address translation natd_interface="xl0" # Set public NAT interface to n1 natd_flags="-f /etc/natd.conf" # Set rules file for the NAT daemonThe second file is /etc/natd.conf which contains the rules for the NAT daemon:
redirect_port tcp 10.0.0.3:22 22 # Redirect SSH packets to the SSH server redirect_port tcp 10.0.0.4:80 80 # Redirect HTTP packets to the HTTP serverThe third file is /etc/rc.firewall which contains the rules for the packet filter. In our case we first divert all IP packets to the NAT daemon. After that, we make sure that IP packets can only go where they are supposed to go. Strictly speaking, we don't need the rules below the one that diverts IP packets to the NAT daemon. However, we would otherwise have to allow all IP packets, and that is something we don't like doing:
# Flush all previous firewall rules /sbin/ipfw -f flush # Divert all IP packets to the NAT daemon /sbin/ipfw add divert natd ip from any to any via xl0 # Allow all IP packets through the loopback interface by the localhost /sbin/ipfw add allow ip from 127.0.0.1 to 127.0.0.1 via lo0 # Allow all IP packets through the public interface (n1) from and to this machine /sbin/ipfw add allow ip from any to 18.104.22.168 in recv xl0 /sbin/ipfw add allow ip from 22.214.171.124 to any out xmit xl0 # Allow all IP packets through the client interface (n2) from and to client machines /sbin/ipfw add allow ip from 192.168.0.1/24 to any in recv rl0 /sbin/ipfw add allow ip from any to 192.168.0.1/24 out xmit rl0 # Allow all IP packets through the server interface (n3) from and to server machines /sbin/ipfw add allow ip from 10.0.0.1/24 to any in recv rl1 /sbin/ipfw add allow ip from any to 10.0.0.1/24 out xmit rl1 # Deny all IP packets not allowed until this point /sbin/ipfw add deny ip from any to any
We want to be able to divert and filter IP packets. The OpenBSD generic kernel has support for both these features, so there is no need to make our own kernel.
The first configuration files to edit are the ones that set the parameters for the network interfaces. File /etc/hostname.ne0 is for the network interface connected to the client LAN (p3):
inet 192.168.0.2 255.255.255.0 NONEFiles /etc/hostname.ne1 and /etc/hostname.ne2 are for the network interfaces connected to the Cable / xDSL modem and NAT machine respectively (p1 and p2). Because we won't be giving them IP addresses, the content of these files is simply:
upNow we should edit the file /etc/bridgename.bridge0 to enable the bridge at startup:
add ne1 add ne2 upThe last configuration file to edit is /etc/ipf.rules which contains the rules for the packet filter. As with the NAT machine, we make sure that IP packets can only go where they are supposed to go. There is a difference however: we want to protect our network from the Internet. We tell the packet filter that it is okay for the NAT machine to send IP packets to the Internet and receive replies on these packets. We also want to offer SSH and HTTP services to the Internet, so we explicitly open these ports on our bridge:
# Allow all IP through the loopback interface by the localhost pass in quick on lo0 from 127.0.0.1 to 127.0.0.1 # Allow all IP through the LAN interface from and to LAN addresses pass in quick on ne0 from 192.168.227.0/24 to 192.168.227.2 pass out quick on ne0 from 192.168.227.2 to 192.168.227.0/24 # Allow all IP through the bridge from the NAT machine pass in quick on ne2 proto tcp/udp from 126.96.36.199 to any keep state pass in quick on ne2 proto icmp from 188.8.131.52 to any keep state # Allow SSH and HTTP through the bridge from the Internet to the NAT machine pass in quick on ne1 proto tcp from any to 184.108.40.206 port = 22 flags S keep state pass in quick on ne1 proto tcp from any to 220.127.116.11 port = 80 flags S keep state # Deny IP not allowed until this point block in quick all
To complete our setup we must configure the clients and servers. Our client LAN is connected through a hub to interface n2 of the NAT machine, so all clients should be in the same subnet as n1. For the same reason, all servers should be in the same subnet as n3. The configuration looks like this:
|IP address||Subnet mask||Gateway|
From one of the client machines we will now start the test of our configuration. Let's see if one of our clients (192.168.0.3) can reach the NAT machine (192.168.0.1):
%ping 192.168.0.1 PING 192.168.0.1 (192.168.0.1): 56 data bytes 64 bytes from 192.168.0.1: icmp_seq=0 ttl=255 time=0.560 ms 64 bytes from 192.168.0.1: icmp_seq=1 ttl=255 time=0.435 ms ...We expect about the same result if we try to reach the NAT machine (10.0.0.1) from one of our servers (10.0.0.3). If this works OK, we should try the PFB. We can do this from the NAT machine by pinging the gateway of our ISP:
%ping 18.104.22.168 PING 22.214.171.124 (126.96.36.199): 56 data bytes 64 bytes from 188.8.131.52: icmp_seq=0 ttl=255 time=20.340 ms 64 bytes from 184.108.40.206: icmp_seq=1 ttl=255 time=21.636 ms ...We can repeat this test from one of our clients or servers to test the NAT daemon. Notice that the Time To Live (ttl) value is 1 lower than before. This is because the NAT machine is now a hop the IP packets have to pass to reach the gateway. Our final test for our setup will be the HTTP server, which we will assume is on IP address 10.0.0.4. First, try to fetch an HTML page from the terminal of the NAT machine itself with this URL:
http://10.0.0.4/directory/file.htmlIf this succeeds, you can try to fetch the same HTML page from a machine on the Internet with this URL:
http://220.127.116.11/directory/file.htmlThe NAT machine should redirect this request to the HTTP server on IP address 10.0.0.4 and give you the same result as the test above.


This section will describe Device Rules in Cacti.
Cacti Automation relies on create device Rules these rules help the automation process figure out what type of device is being scanned. Based on the device rule REGEX the automation engine can apply the appropriate template to the device i.e if there is a REGEX for IOS. The automation engine may pick the Cisco router template for the device.
Below you will be able to choose a REGEX that will help identify the device the REGEX can be in any Perl supported format. You can select either to match by system description i.e the device hostname or the system name i.e Cisco.
An example regex for Cisco would be (Cisco Internetwork Operating System Software|IOS) This matches based on the system name of the Cisco IOS software.
Using this info automation would apply the appropriate device template you choose. If you see below we match for a system that matches Linux once a device is found with that system name it is assigned the Net-SNMP device template.
If the System/Device you are matching has a unique OID you can also match based on that.