Polycom 7000 manual For SIP calls gatewayed to an

Call Server Configuration

Field	Description
For SIP calls gatewayed to an	If this option is selected, when the system uses dial rules to attempt to resolve
external gatekeeper, use the	a SIP call to an external gatekeeper, the Call Server sets the destination in the
H.323 email ID as the destination	LRQ message to the H.323 email ID (such as 1234@example.com) rather
	than utilizing the E.164 number alone (such as 1234).
	Some external gatekeepers, such as the RealPresence Access Director
	system, may need the additional domain information in the LRQ message to
	correctly resolve the LRQ request.
	If this option is off, SIP calls gatewayed by the RealPresence DMA system to
	a RealPresence Access Director configured as an external H.323 gatekeeper
	fail because the gatekeeper doesn't have enough information to route the call.
	Note: This option affects communications with all external H.323 gatekeepers
	to which the RealPresence DMA system gateways SIP calls.
SIP Settings
Minimum SIP registration interval	The minimum time between “keep alive” messages to SIP endpoints.
(seconds)	Must be less than or equal to the registration refresh interval and in the range
	150-3600.
SIP OPTIONS ping timer	The frequency with which the system sends SIP OPTIONS requests when no
(seconds)	other SIP traffic is received from the SIP peer.
	Must be in the range 1-10000. The default value is 10.
SIP OPTIONS ping failure status	Specifies which responses to the OPTIONS request indicate that a SIP peer
codes	is not responsive.
	Valid input is a comma-separated list or dash-separated range of three-digit
	numeric codes; an empty field is acceptable as well.
	The default value is 503.
SIP max breadth	The maximum number of SIP peers that will be tried at once.
	This option applies when the Routing policy for a dial rule with the action
	Resolve to external SIP peer is set to All in parallel (forking).
	Must be in the range 1-99. The default value is 60.
Try next SIP peer timeout	The timeout in seconds when sending a SIP OPTIONS ping or an INVITE to a
(seconds)	SIP peer. This value can be a numeric value in the range 0.1-31.0.
	The default value is 5.0.
SIP peer dial rule timeout	The number of seconds after invoking the dial rule that the dial attempt is
(seconds)	cancelled.
	Must be in the range 1-300. The default value is 25.
Nonresponsive SIP peer status	Specifies which responses to an initial SIP INVITE indicate that a SIP peer is
codes	not responsive.
	Valid input is a comma-separated list or dash-separated range of three-digit
	numeric codes; an empty field is acceptable as well.
	The default value is 503.

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