Abstract: There are many software tools for designing digital filters. For example, the MATLAB application package has the ability to model and calculate different types of digital filters. However, little attention is paid to frequency sampling filters (FSF) and there are no well-known and available software tools for their calculation. Such filters have high computational efficiency and flexibility in reconfiguring the frequency bands that are analyzed. They can be useful in the design of high-performance spectrum analyzers, speech signal processing, quadrature phase signal manipulation, etc. The frequency sampling method is one of the methods of designing filters with finite impulse response (FIR). It consists in obtaining the coefficients of the transfer function of the filter directly from the samples of a given amplitude-frequency characteristic (frequency response). This characteristic is easily set for stopbands (it is zero) and passband (it is one). Therefore, the weighting coefficients of the transfer function, determined by the frequency response in the passband are equal to one. The resulting frequency response of the filter differs from the desired at intervals between interpolation nodes. This difference is significantly determined by the specified values of the coefficients in the transition bands (so-called transition coefficients). Thus, the main problem of approximation is contained in the correct choice of the order of the transfer function and the optimal determination of the transition coefficients. The purpose of this article is to describe the capabilities of the software tool “Designer of bandpass frequency sampling filters”, which allows you to optimally solve the problem of approximation FSF, and demonstrate an example of solving such a problem. The article has a scientific and methodological nature.