“VoIP stands for Voice over Internet Protocol and allows you to make telephone calls using the Internet or other computer networks. VoIP converts the voice signal from your telephone into a digital signal that travels over the internet then converts it back at the other end so you can speak to anyone with a regular Telephone.”

“SIP stands for ‘The Session Initiation Protocol’ and is an application-layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

It is widely used as signalling protocol for Voice over IP, along with H.323 and others. SIP is addressing neutral, with addresses expressed as URL/URL’s of various types, such as H.323 address, E.164 telephone numbers or email like addresses.”

“H.323 is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network.

It is currently implemented by various Internet real-time applications such as NetMeeting and Ekiga (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over ISDN, PSTN or SS7. H.323 is commonly used in Voice over IP (VoIP, Internet Telephony, or IP Telephony) and IP-based videoconferencing. Its purpose is thus similar to that of the Session Initiation Protocol.”

“IAX is the Inter-Asterisk eXchange protocol used by Asterisk, an open source PBX server from Digium. It is used to enable VoIP connections between Asterisk servers, and between servers and clients that also use the IAX protocol.

IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated almost universally in favour of IAX2.”

“Use a Number to Terminate to an Asterisk Server so PSTN and Mobile Callers will be able to Call you.”

“Number Terminates to a SIP Phone/Address so PSTN and Mobile Callers will be able to Call you.”

“Virtual Chat Service.”

“DTMF stands for Dual Tone Multi Frequency and is used for telephone signalling over the line in the voice-frequency band to the call switching centre. The version of DTMF used for telephone tone dialling is known by the trademarked term Touch-Tone, and is standardised by ITU-T Recommendation Q.23. Other multi-frequency systems are used for signalling internal to the telephone network.”

“Yes, you can select whether to Route Calls with RFC2833 DTMF Signalling or Inband DTMF in the Novero Platform. When you set up Call Routing to a SIP Address just either RFC2833 or Inband DTMF.”