Is it possible to combine two CFU455IT +/-2kHz (4kHz) ceramic filters to get a +/-1.25kHz bandwith (2.5kHz) filter?
Yes, but no.
Well, you kind of can, but you really can't:
When you just put two resonant filters in series, like these ceramic filters, you do, in fact, get a sharper resonance in passband; will it have a  2.5 kHz wide 3dB-bandwidth? Only with extreme luck; your datasheet doesn't specify.
It's a technique actually used with crystal resonators, usually, in the form of ladder filters, but when you build these, you'd typically be very careful to select resonators that are very well-matched; otherwise, you'd be running the risk of strong insertion loss, or generally surprising passband shape.
What to do instead
Think about it this way:
It might simply not be worth it building an analog bandpass filter, as you'd never need that. Which system of yours really requires a 2.5 kHz bandwidth on a 455 kHz IF?
The fact alone that you're planning to work on a 455 kHz band shows you've thought about filtering architecture: That's a classical intermediate frequency, for which narrow filters are available – and with narrow, we mean &quot;ratio of bandwidth to center frequency is small&quot;; it's, due to pure physics, way harder to build a 5 kHz filter around 1 GHz than around 455 kHz.
Superhet, active filters
So, if you really need less bandwidth, the &quot;natural&quot; way would be to add another superhet stage to go to an even lower IF, say, 8 kHz (or really anything above 2.5 kHz, I just materialized that number out of thin air) and look into active multi-stage filters; Analog Devices has an excellent Filter Design Wizard that takes you through all the design steps and delivers a design, BOM, curves including error margins.
Unless masochistic, don't do analog filters
Then, again, I'd personally really not do that, for 99.95% of use cases.
You've got a 4 kHz wide passband around 455 kHz after one stage of filtering. Because I'm lazy with the arithmetic, let's say that's 5 kHz bandwidth that you'd have to deal with.
So, with a microcontroller including an ADC that can do 800 thousand samples per second (or more), you'd be oversampling¹ by a factor of ; that means you gain 80/4 = 20 bits in ADC resolution. At which point even a terribly performing 10 bit ADC absolutely suffices to reconstruct even the weakest signal out of noise, and possible interferer within the 1.5 kHz bandwidth you wanted to cut off with your second ceramic filter.
All you'd need is

a microcontroller, ideally one that includes
an ADC, as well as,
an ADC buffer (i.e. a unity-gain amplifier, just so that the ADC's input impedance doesn't affect the upstream analog signal chaing; typically, you want this, but if you've already got an IF amplifier, potentially not necessary),
as well as a bit of firmware code that

sets up the ADC,
configures some DMA transfers to get the memory from the ADC when a conversion is done automatically,
sets up a timer so that ADC conversion is done every \$\frac{1}{800\,\text{kHz}\$ automatically, and
processes the resulting digital signal.



Such microcontrollers (example) aren't cheap (example costs: 4.72€ in single-unit quantities); but compared to the cost and complexity of an extra superhet stage, or even better resonant filters at 455 kHz, not actually!
Once you've got these digital samples, you can do any filtering you'd do in analog domain, but digitally, i.e. without component variations, without temperature dependency, without needing to hand-tune passive components until things are &quot;just right&quot;...
You can either do that on the microcontroller itself (given it's fast enough to do the computation), or you could off-load the processing to an application processor (e.g. the CPU in your laptop, or the CPU in your Raspberry Pi, although the first generations of RPis are really terrible computation platforms), e.g. by sending the samples over USB.
A realistic approach here would be to let the firmware:

Get the samples from the ADC into a buffer using DMA,
run a relatively relaxed half-band filter²
which decimates (i.e. throws away \$N-1\$ of \$N\$ samples) by a decimation factor \$N=2\$
and offers the remaining half-ADC-sample-rate sample stream via Full-Speed USB2 (i.e. 12 Mbit/s) to a PC,

where the PC then

further filters and decimates down to say, 24 kHz containing the signal of interest,
then uses your favourite frequency recovery (e.g. a PLL, depends on the kind of RF signal you're dealing with!) to home in very exactly on the signal of interest, so the analog side (oscillators, filters) doesn't have to be that frequency-exact
further filters down to an arbitrary, adjustable bandwidth on the fly,
applies whatever signal processing is relevant for your signal of interest (AM demod? FM demod? DRM demod? A PSK receiver? FreeDV demod? …) and generates the relevant output.

For the PC side of things, do have a look at GNU Radio, it's meant to be a builder toolkit for exactly such signal processing chains.

¹ The minimum sampling rate necessary to unambigously represent a &lt;5 kHz wide signal is 10 kHz; if your sample rate \$f_s\$ is larger than that, you're oversampling by a factor \$k=\frac{f_s}{10\,\text{kHz}}\$.
² When actually using 800 kSamples/s, 455 kHz would be aliased to 345 kHz, so you'd use a half-band high-pass filter, very possibly a CIC, which avoids having to do any multiplications, only additions and sign inversions, so you can keep working on the fixed-point numbers your ADC gives you without potentially getting into arithmetic trouble; it's computationally usually advantageous to do that.