Configuring Asterisk as SIP Gateway Server

My plan is to make outgoing calls and get incoming calls thru SIP s/w phone.

For that in implementation we choose open-sips server. We have installed open-sips with help of

" "

As the next step, I understood that incoming calls needs DID number.

Moreover for outgoing calls OpenSIPS needs asterisk for PSTN connectivity. For that :

[b]If Gateway server has to be configured from our end, hope we need to install asterisk in a separate server and

integrate asterisk with opensips[/b] basing the link " "

Will this be enough for making outgoing calls.

If you use Asterisk to support outgoing PSTN calls, you do not need to buy, what is mis-named in the Asterisk world, a DID service. Asterisk can accept incoming PSTN calls directly, as well as making outgoing ones. The incoming calls may or may not use Direct in Dialing, depending on whether you need callers to be able to access extensions without going through an automated attendant.

I’m afraid i’m not familiar with Open SIPS, so I can’t help with the rest.