Cisco Cisco IOS Software Release 12.2(4)T

MGCP optionally supports multiple call agents, which can eliminate a potential single point of failure 
in the voice network.

The Cisco IOS CLI allows you to set up MGCP on the gateway and to identify the Cisco CallManager. 
Cisco CallManager assumes control over establishing and tearing down connections between IP 
endpoints on your network and endpoints connected through the Public Switched Telephone Network 
(PSTN).

MGCP uses User Datagram Protocol (UDP) for establishing audio connections over IP networks. 
However, MGCP also uses hairpinning to return a call to the PSTN when the packet network is not 
available. 

Creating a call connection involves a series of signals and events that make up the connection process. 
The signals might include such indicators as the off-hook status, a ringing signal, or a signal to play an 
announcement. These events and signals are specific to the type of endpoint involved in the call. 

MGCP groups these events and signals into packages. A trunk package, for example, is a group of events 
and signals relevant to a trunking gateway, and an announcement package is a group of events and signals 
for an announcement server. 

In an MGCP-enabled gateway, the out-of-band dual tone multi-frequency (DTMF) package is loaded 
upon startup. Once the out-of-band DTMF capabilities are configured in the Cisco CallManager MGCP 
gateway user interface, the router sends symbols over the UDP control channel to represent any DTMF 
tones it receives. Cisco CallManager interprets these symbols and passes on the DTMF signals, out of 
band, to the signaling endpoint. 

The Cisco 2600 and 3600 Routers MGCP Voice Gateway Interoperability with Cisco CallManager 
feature includes the following:

When you are using MGCP with a Cisco 2600 series or 3600 series router, all configuration elements 
associated with dial-plans are controlled by Cisco CallManager and should not be configured in the 
Cisco 2600 series or 3600 series gateway for MGCP-managed endpoints. 

Supplementary services includes call hold, call transfer when the line is busy or there is no answer, call 
forwarding, and three-party call conferencing to and from the PSTN or a private branch exchange (PBX).

Call hold is a function that places the handset into mute mode. Both the transmitter and receiver function 
are disengaged for a period of time until the hold button is pressed again to reconnect the parties. 

Call transfer is a function that transfers a call to a third party through a pre-programmed button that 
performs the hookswitch and draw with what is called the recall dial tone. The receiver of the call then 
dials the third-party number, waits for the line to ring and for the new called party to answer, and then 
hangs up.

Call forwarding is a function that allows you to forward calls dialed from the original location to a 
remote location within or across the network.

Three-party call conferencing is similar to the transfer function, but rather than transferring the call to a 
third party, the third party called is added to the call. The conference feature allows the three parties to 
converse without worrying about cutting each other off.