[Asterisk-doc] Sip server

["Juan G. Castañeda"]

Sirs:

1.-  Can clients of a SIP server such as SER, use an Asterisk  SIP 
gateway to give connectivity to the PSTN network?

2.-  Conversly, Calls that originate in the PSTN can traverse an 
Asterisk SIP gateway to terminate at a SIP endpoint?

3.-  Finally,  may a call originate and terminate in the PSTN, but cross 
a SIP Sever-based network in the middle?

Could you please explain?

Regards,

Juan G. Castañeda