I'm currently working on the integration of one customer.
They already have an asterisk server and the first step is to link his asterisk to our MTE asterisk with a SIP TRUNK and we just forward calls through our PSTN access

is there a way to specify MTE that everything coming from SIP TRUNK ID1, forward all to PSTN access ?
I was able to do so by the recall script but only with a dedicated DID
so I don't want to create incoming route for all numbers he is going to call ?

Thank you in advance