In VoIP network with Asterisk being the server or SIP proxy the secure calling can be achieved by enabling TLS to encrypt the signalling and enabling SRTP or ZRTP to encrypt the media or data/voice. Once implemented SIP UA, softphone or IP phone, can be set to use TLS instead of UDP or TCP as it’s transport. The advantage of choosing TLS is that the SIP traffic exchanged between SIP UA and Asterisk will be encrypted, it means it will take a considerable amount of time and effort for the Man in The Middle to decrypt it without the encryption key, if not possible.
Below are screenshots of CSipSimple (Free and Open Source SIP softphone on Android) calling the other party and having end-to-end encryption during the call with Asterisk as the server:
