Hello,
Im new in asterisk, I succesfully created a SIP and a Dahdi trunk on my system. I can receive calls from the PSTN and make calls to the SIP trunk of my partner.
Now I have the requirement of route all incoming calls from the PSTN trunk to the SIP trunk.
Can somebody give me an idea of how to accomplish this?
Thanks in advance,
Assuming that the DAHDI trunk is an analogue one, or otherwise doesn’t have dialed digits supplied, use DISA, Read, or WaitExten. The main thing is to make sure that the resulting numbers are interpreted in a context that doesn’t have the ability to make toll calls. Normally your local phones would be in a context that included that context.
