Convolution does not change the sample rate, but the sample rates of the two inputs need to match. A simple way to do this in your case is to zeropad the spectrum of your signal by a factor of 2 and the spectrum of the channel impulse response by a factor of 25. That way they will both have a sample period of 0.005.
EDIT: There is nothing wrong with this answer. Zeropadding the spectrum of a signal is a simple and common way to change the sample rate. Below is an example using octave.
% FFT function with swapping.
sfft = @(x) fftshift(fft(ifftshift(x)));
sifft = @(x) fftshift(ifft(ifftshift(x)));
% The frequency sample locations for a given
% sample rate and DFT size.
fvec = @(N,fs) (0 : fs/N : (fs-fs/N)).' - (fs - mod(N,2)*fs/N) / 2;
% Parameters for the input signal.
fs1 = 100; % Sampling frequency (Hz)
T1 = 100; % Signal duration (s)
t1 = 0 : 1/fs1 : T1; % Sample times (s)
N1 = length( t1 ); % Number of input samples
B1 = 0.8 * fs1; % Signal bandwidth (Hz).
f1 = fvec(N1,fs1); % Frequency sample locations (Hz)
% Define the some bandlimited random signal.
x1 = sifft( exp(1j*2*pi*rand(N1,1)) .* ( abs(f1) <= B1/2 ) );
% Parameters for the channel impulse response.
fs2 = 8;
T2 = 20;
t2 = 0 : 1/fs2 : T2;
N2 = length( t2 );
B2 = 0.8 * fs2;
f2 = fvec(N2,fs2);
% Some random numbers to represent the channel response.
x2 = sifft( exp(1j*2*pi*rand(N2,1)) .* ( abs(f2) <= B2/2 ) );
% To convolve the input signal with the channel
% impulse response, they both need to be at the
% same sample rate. We do this by zeropadding the
% spectrums of each such that both are sampled
% at the same rate. We use the least common multiple
% so that the zeropad factor is an integer.
fout = lcm( fs1, fs2 );
% The number of samples in the zeropadded spectrums
% for both the input signal and the channel response.
Nzp1 = N1 * fout / fs1;
Nzp2 = N2 * fout / fs2;
% The indices that correspond to the lower
% frequencies. These are the bins into
% which we copy the specrtums.
zinds1 = fvec(N1,N1) + floor( Nzp1/2 ) + 1;
zinds2 = fvec(N2,N2) + floor( Nzp2/2 ) + 1;
% Insert the spectrum into an array of zeros
% and inverse FFT to get the signal at the
% common sample rate.
tmp = zeros( Nzp1, 1 );
tmp( zinds1 ) = sfft( x1 );
x1_out = sifft( tmp );
tmp = zeros( Nzp2, 1 );
tmp( zinds2 ) = sfft( x2 );
x2_out = sifft( tmp );
% Now that both signals have the same sample rate,
% they may be convolved.
y = conv( x1_out, x2_out );
% The frequency sample locations of the convolution.
fy = fvec( length( y ), fout );
plot( fy, abs( sfft( y ) ) );
xlabel( 'Frequency (Hz)' );
ylabel( 'Amplitude' );
title( 'Spectrum After Convolution' );
xlim( [ -10 10 ] );