Interfacing PSTN with a legacy PBX

I have two questions from Asterisk experts:
Is it possible to interface PSTN with a legacy PBX using an asterisk powered server?
if yes, assuming an E1 ISDN-PRI Iine (30B+D), is it possible to change/modify signalling information by this asterisk server?

Should have been in the support forum.

The answer is a definite maybe.

By definition, any legacy paBx should be able to interface with the PSTN without adding a tandem switch, so there must be subtleties in your requirement that you haven’t detailed. Asterisk support many circuit switched interfaces, including your E1, but whether or not it can do what you want without code modification, can only be answered with a much more detailed specification of the requirements.

As it is open source, modification of the code is also a possibility.

If you are thinking of using Digium supplied hardware for your E1 interface, you should be able to get information from Digium Sales.

Thank you for reply.

A simple example of what I meant is ability to remove DTMF/MF on media channels or monitor the bandwidth usage on these channels (and probably block the channels that exceed a predefined usage threshold) . Or consider the case when we like to restrict call destinations and send tear down signalling message when an illegal call (to a forbidden number) is detected.

The required hardware would be at least an asterisk based card (maybe digium) equipped with two E1 interfaces (One to PSTN and ONE to PBX). It is not possible to integrate this module with PBX and it should work transparently as if the PBX is connected directly to PSTN.

How much of it can be borrowed from open source? Is it straightforward for asterisk based cards? The alternative is to develop the application on more expensive cards that make use of dedicated hardware/API.

Sounds like you are trying to find a technical solution to a social problem - that rarely works well.

Asterisk would give you extreme flexibility whilst it was still in the call setup phase. I don’t believe there is any mechanism to notch filter DTMF tones on circuit switched connections, although you might be able to misconfigure a VoIP one to achieve this.

It might be possible to use call features to hijack in call use of DTMF, although that is not their primary purpose.

I thought it (DTMF clamping) could work when our interface acts like a “back end” from both ways (PSTN and PBX).
Can we remove it this way?


I’m guessing you are not from a telecoms background.

Why would you want to remove DTMF ?
An ISDN call is 64K if by “channels that exceed a predefined usage threshold” you mean limit the number of channels being used.
and as for “send tear down signalling message when an illegal call” this is called call baring and handled by the dial-plan.

It sounds like you want to add a system to handle call barring and a standard asteris server will handle all this. Just in stall a 2 port card and design a dialplan,

Thank you very much Ian.
It is almost what you described. I just want to make sure it is possible to imagine something more than dial-plan. In simple words, an ISDN firewall.
Dial plan is certainly a part of it. But it would be great if one could monitor media/signalling channel activity. Like a data firewall, one may want to set rules on the content of media channels, discriminate call type not only by means of “bearer capability” but also B-channel analysis. Is it possible?