If you need to separate the SIP server funcionality (registration, proxy, etc.) you shouldn’t use Asterisk. Since * offers such a unified treatment (SIP, POTS, etc., etc.) of telephony technologies, it also hides and “bundles” some of the inside details. With Asterisk you are getting a heck of a lot more than SIP service.
To get a SIP server (and only a SIP server) you should look into the SIP Express Router, which comes in two almost identical versions: ‘ser’ and ‘openser’ because their developers had a fight and parted ways.
You must be asking about “Zapata” (not Zapaca). If that’s the case, you can get started without telephony hardware BUT you MUST get a voice card of some sort (Digium is the path of least resistance) when you go into actual use and production.
sorry for the typo - zapaca, haha.
you know, I knew * is a unified/integrated s/w, and currently I need only
its SIP capablity, but in the future I will use more of its function.
so, how can I set my * to connect other SIP server ? (then, when I dial
into my * box, when press “2”, the * will connect into other SIP server
with the pre-define user/pass.)