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Call Server Configuration
Polycom, Inc. 236
For SIP calls gatewayed to an
external gatekeeper, use the
H.323 email ID as the destination
If this option is selected, when the system uses dial rules to attempt to resolve
a SIP call to an external gatekeeper, the Call Server sets the destination in the
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
(seconds)
The minimum time between “keep alive” messages to SIP endpoints.
Must be less than or equal to the registration refresh interval and in the range
150-3600.
SIP OPTIONS ping timer
(seconds)
The frequency with which the system sends SIP OPTIONS requests when no
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
codes
Specifies which responses to the OPTIONS request indicate that a SIP peer
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
(seconds)
The timeout in seconds when sending a SIP OPTIONS ping or an INVITE to a
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
(seconds)
The number of seconds after invoking the dial rule that the dial attempt is
cancelled.
Must be in the range 1-300. The default value is 25.
Nonresponsive SIP peer status
codes
Specifies which responses to an initial SIP INVITE indicate that a SIP peer 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.
Field Description