Polycom 1500/1800/2000/4000 manual Calling a SIP Endpoint in a Remote Domain

Gateway Calls

When the flag is set to NO (default), if the Gatekeeper is not functioning, calls from endpoints will be connected directly to the Entry Queue, Conference or Meeting Room that was dialed.

The method by which dial-in calls are accepted or rejected is dependant on the flag value and the availability of the Gatekeeper as summarized in the following table.

GK_MANDATORY_FOR_CALLS_IN - System Flag

Flag Value	Gatekeeper Available	Results
NO	NO	Dial-in call is connected bypassing the Gatekeeper.
NO	YES	Dial-in call is connected using the Gatekeeper.
YES	NO	Dial-in call is rejected.
YES	YES	Dial-in call is connected using the Gatekeeper.

Enabling or Disabling Direct IP Dialing

The direct IP dialing is enabled by default. To disable it, manually add the flags

GK_MANDATORY_FOR_CALLS_OUT and GK_MANDATORY_FOR_CALLS_IN to the System Configuration - MCMS_PARAMETERS dialog box and for each flag enter the required value (YES or NO).

For more information on flag definition, see Modifying System Flags.

For flag changes (including deletion) to take effect, reset the Collaboration Server. For more information see Resetting the Collaboration Server.

Calling a SIP Endpoint in a Remote Domain

The Gateway functionality allows the connection of H.323 and SIP endpoints to SIP endpoints residing in a remote domain, different to that of the MCU’s domain.

This functionality can be enabled when:

●The calling endpoint may be H.323 or SIP.

●The destination endpoint must be SIP.

●The dial string from the calling endpoint to the Collaboration Server includes the gateway dial-out number of the SIP endpoint that is located in the remote domain.

By definition, a dial-in string cannot contain two domains and it is therefore necessary that the dial string be of the following format:

mcu-meeting-room*dest%40dest-domain@mcu-domain

Where:

●The domain of the gateway call is mcu-domain

●The remote destination domain is dest%40dest-domain.

●The “%40” is replaced with an “@” when the gateway call is dialed to the remote SIP destination.

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